A hearing device

By employing multiple input converters and adaptive beamforming filter units in the hearing aid, and updating the beamforming weights using feedback estimates, the microphone-to-acoustic receiver feedback problem is solved, thereby improving the acoustic gain and hearing correction performance of the hearing aid.

CN115119125BActive Publication Date: 2025-02-28OTICON
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Patent Information

Application Number
CN202210480038.8
Authority / Receiving Office
CN · China
Patent Type
Patents(China)
Current Assignee / Owner
Priority Date
2018-02-09
Filing Date
2019-02-09
Publication Date
2025-02-28
Estimated Expiration
2039-02-09

AI Technical Summary

Technical Problem

In existing hearing aids, feedback issues between the microphone and the acoustic receiver limit acoustic gain, resulting in limited hearing correction performance, especially in small fully implantable hearing aids (CIC).

Method used

Multiple input converters and adaptive beamformer filtering units are employed. The beamformer weights are updated through feedback estimates. The MVDR beamformer and generalized sidelobe canceller structure are used to achieve adaptive reduction of feedback.

Benefits of technology

It effectively reduces feedback, increases the acoustic gain of hearing aids, and improves auditory correction performance, especially significantly improving auditory perception in small hearing aids.

✦ Generated by Eureka AI based on patent content.

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Abstract

The present application discloses a hearing device, which comprises an ITE part, wherein the ITE part comprises: a shell configured to be at least partially located in the ear canal of a user, at least three input transducers for providing corresponding electrical input signals, an output transducer, and a beamformer filter unit connected to the at least three input transducers and the output transducer; wherein the beamformer filter unit comprises: a first beamformer for spatially filtering sounds in the environment based on electrical input signals from the at least two input transducers facing the environment; and a second beamformer for spatially filtering sounds reflected from the eardrum based on at least one electrical input signal from the at least one inner input transducer facing the eardrum and at least one of electrical input signals from the at least two outer input transducers facing the environment.
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Description

[0001] This application is a divisional application of Chinese patent application 201910109342.X filed on February 9, 2019, with the invention name “Hearing device including a beamformer filter unit for reducing feedback”. Technical Field

[0002] The present application relates to the field of hearing devices such as hearing aids, and in particular to feedback from an output transducer to an input transducer of a hearing device. Background Art

[0003] EP2843971A1 discloses a hearing aid device, comprising an "open fitting" providing ventilation, a receiver disposed in the ear canal, a directional microphone system comprising two microphones disposed in the ear canal and on the same side of the receiver, and a device for canceling acoustic feedback based on sound signals detected by the two microphones. Improved feedback reduction can thereby be achieved while enabling a relatively large gain to be applied to the incoming signal.

[0004] In state-of-the-art hearing aids, omnidirectional microphones are known to provide satisfactory audiological performance for very small hearing instruments that are located almost invisibly in the entrance to the ear canal. It is also known to have slightly larger hearing aids with microphones that are placed further out in the ear or behind the pinna, with increased audiological performance being obtained from the use of directional microphone systems. Such directional systems are able to distinguish between sounds coming from the front area from the hearing aid user's perspective and sounds coming from other directions in the horizontal plane. Therefore, from a traditional point of view, CIC hearing instruments have only one microphone and larger ITEs usually have two microphones together for directional performance.

[0005] Both small CICs and larger ITE hearing instruments have limited acoustic gain from the incoming sound at the microphone to the acoustic receiver output. This gain is limited by feedback problems caused by unwanted signals passing from the receiver back into the microphone. This problem can be mitigated by anti-feedback systems based on feedback path estimation. This is well known. Summary of the invention

[0006] Hearing Devices

[0007] In one aspect of the present application, a hearing device such as a hearing aid is provided, which is configured to be located at or in the ear of a user or fully or partially implanted in the head at the ear. The hearing device comprises:

[0008] - a plurality of input transducers for providing respective electrical input signals representative of sounds in a user's environment;

[0009] - an output transducer for providing a stimulus perceivable as sound by a user based on said electrical input signal or a processed version thereof;

[0010] - an adaptive beamformer filter unit connected to the input unit and the output unit, configured to provide a spatially filtered signal based on a plurality of electrical input signals and adaptively updated beamformer weights;

[0011] - A feedback estimation unit providing a feedback estimate of a current feedback path from said output transducer to each of said input transducers.

[0012] The hearing device is configured such that at least one of said adaptively updated beamformer weights of an adaptive beamformer filtering unit is updated in dependence on said feedback path estimate.

[0013] Thereby, a hearing device comprising an alternative feedback reduction system may be provided.

[0014] The plurality of input transducers may be or include microphones. The beamformer filter unit may constitute or include an MVDR beamformer (MVDR=minimum variance distortionless response). The term stimulus perceivable as sound in this specification primarily means a stimulus which may result in feedback to the input transducer. When electrical stimulation is applied alone (e.g. in a cochlear implant) the feedback problem does not exist, but feedback may occur when there is a combination of electrical and acoustic stimulation (e.g. so-called bimodal fitting).

[0015] The hearing device may be configured to provide each of the respective electric input signals as a subband signal X in a time-frequency representation (k, m) i (k,m), i=1, ..., M, where M is the number of input transducers, where k and m are frequency and time indices respectively, and where k=1, ..., K. The hearing device may comprise an analysis filter bank to provide a given electrical input signal in a time-frequency representation. In an embodiment, each input path from the M input transducers comprises an analysis filter bank. The analysis filter bank may comprise a Fourier transform algorithm such as a short-time Fourier transform (STFT) algorithm, which provides a subband signal in a time-frequency representation (m, k), where each time frame (m) comprises K time-frequency units (e.g. STFT windows), each time-frequency unit comprising a complex value of the subband signal corresponding to a particular frequency index k at the time m concerned. The hearing device may comprise a synthesis filter bank for converting the electrical signal in the subband (or time-frequency) representation into a time domain signal. The hearing device may comprise at least one synthesis filter bank (for hands-free telephony or binaural communication, other synthesis filter banks may be necessary).

[0016] The adaptive beamformer filter unit may comprise a first set of two (e.g. mutually orthogonal) beamformers:

[0017] a) a (first) beamformer C1 configured to leave the signal from the target direction (substantially unchanged); and

[0018] b) a (second) (e.g. orthogonal) beamformer C2 configured to (substantially) cancel signals from the target direction; and

[0019] The adaptive beamformer filter unit is configured to provide a synthesized directional signal Y(k)=C1(k)-β(k)C2(k), wherein β(k) is an adaptive factor for determining the adaptive update of the adaptively updated beamformer weights, wherein β(k) is determined based on the feedback estimate. The adaptive factor β(k) can be determined from the following expression:

[0020]

[0021] where k is the frequency index, * refers to the complex conjugate, and <·> refers to the statistical expectation operator, and c is a constant, and (C F1 ,C F2 ) constitutes a second set of beamformers applied in the frequency domain to the feedback path estimate.

[0022] The terms "substantially" ("substantially not modified" and "substantially cancel", respectively) used in conjunction with the first and second beamformers are used to indicate possible minor deviations from the ideal properties of the beamformers in question. It is generally not possible (at all frequencies) for the beamformers in question to completely cancel signals from a specific direction because of physical imperfections of actual implantations of specific hearing devices.

[0023] It should be noted that the "target direction" can be viewed as a specific direction such as the front direction (for hearing aid users) or the direction of one's own voice (for headphone applications). Alternatively, the "target direction" can be interpreted as a set of beamformer weights that attenuate a range of directions, such as distracting noise. Figure 1A This is particularly relevant in the configuration shown in , where the "target direction" can be considered as all external sounds. The noise is thus minimized under the constraint that the signal from the target direction is not changed. <·> refers to the averaging of the signal, for example by a first-order IIR low-pass filter (in Figure 2 and 4 In contrast to the adaptive beamformer, which cancels out external noise, we expect the “noise” (i.e. feedback) to be low in the current setup (see Figure 4 ). Thus, we have the advantage of a slower adaptation / adjustment (longer time constant). If we detect a change in the feedback path, it would be an advantage if the time constant could be reduced (faster reaction) as soon as a change in the feedback path has been detected.

[0024] The advantage of the discussed beamformer structure (Y=C1-βC2) is that the factor β responsible for noise reduction is multiplied only onto the second (target cancellation) beam pattern C2 (so that the signal received from the target direction is not affected by any value of β). The constraints of this minimum variance distortionless response (MVDR) beamformer are built into the characteristics of the generalized sidelobe canceller (GSC) structure.

[0025] As described in EP3253075A1, β(k) can be determined directly from the noise covariance matrix (which is obtained from the input signal (e.g. via the feedback path estimate) and the beamformer weights) without the step of calculating the fixed beamformer. This may be advantageous in cases where the fixed beamformer weights can vary. In other words, we can determine β directly from the signal (here for the two-input case)

[0026]

[0027] where x represents the electrical input signal such as a microphone signal ( Figure 2 (X1,X2)) or the feedback estimator ( Figure 4 In As an alternative, we can get the noise covariance matrix C from v Determine β, that is

[0028]

[0029] where w C1 =(w 11 (k),w 12 (k)) T is a vector representing the first beamformer (C1) comprising a first set of frequency-dependent complex weighting parameters, and w C2 =(w 21 (k),w 22 (k)) T is a vector comprising a second set of frequency-dependent complex weighting parameters representing the second beamformer (C2). This may be an implementation choice. It should be emphasized that the noise covariance matrix C v It can be obtained from the feedback estimator:

[0030] C v = <FF H >

[0031] in

[0032]

[0033] Or, as an alternative, expressed as

[0034]

[0035] in T Refers to transposition, H refers to transpose and complex conjugate (and * refers to complex conjugate), and <·> refers to time averaging (eg, equivalent to low-pass filtering, such as implemented by an IIR filter).

[0036] Instead of an absolute feedback path estimate from the output transducer to each input transducer, a reference input transducer may be selected, and the absolute feedback path determined with respect to the reference input transducer, and the relative feedback paths from that input transducer to the remaining input transducers. Thus, updating of the feedback path estimates may be simplified.

[0037] The advantage of using the feedback path estimate over the microphone signal is that the updating of the adaptive beam pattern will be less affected by external sounds (see Figure 1A ).

[0038] The first group (eg, two mutually orthogonal) of beamformers (C1, C2) may be fixed beamformers. The first group (eg, two mutually orthogonal) of beamformers (C1, C2) may be adaptively determined.

[0039] The second group of beamformers (C F1 ,C F2 ) may be fixed beamformers. In an embodiment, the second group of beamformers (C F1 ,C F2 ) Adaptively determined.

[0040] The second group of beamformers (C F1 ,C F2 ) may have the same weights (w) as the first set of beamformers (C1, C2) 11 ,w 12 ),(w 21 ,w 22 ), but can be estimated from the feedback path In other words,

[0041]

[0042] in represents the feedback estimator (see Figure 4 An exemplary two-microphone embodiment of

[0043] Hearing devices may include

[0044] - comprising a first set of frequency-dependent complex weighting parameters w representing said first beamformer (C1) 11 (k),w 12 (k) memory;

[0045] - comprising a second set of frequency-dependent complex weighting parameters w representing said first beamformer (C2) 21 (k),w 22 (k) memory;

[0046] - where the first and second sets of weighting parameters w 11 (k),w 12 (k) and w 21 (k),w 22 (k) Predetermined, for example as an initial value, which may be updated during operation of the hearing device.

[0047] The memory may be implemented as one memory or as separate memories. The memory may, for example, form part of a processor or any other functional unit.

[0048] The number of sets of predetermined feedback path estimates may correspond to a specific acoustic situation and may be stored in a memory of the hearing device for each of the plurality of input transducers. In an embodiment, a plurality of different predetermined feedback paths are stored in a memory of the hearing device. Depending on the specific feedback situation, an appropriate feedback path may be selected and used to determine the adaptive beamformer weights β(k).

[0049] The adaptive beamformer filter unit may include a plurality of different fixed beamformers, which may be switched depending on the acoustic situation.

[0050] As an alternative or in addition, the hearing device may be configured to control the adaptation rate of the feedback estimation unit (algorithm) based on the "distance" between corresponding reference feedback paths (e.g. the Euclidean distance between the amplitude and / or phase or the logarithm of these at different frequencies) and the current feedback path estimate. Thus, as long as the current feedback path estimate is close to one of the reference feedback estimates, a relatively slow adaptation may be applied. The "adaptivity" of the beamformer is mainly related to β (see Figure 4 ). However, the fixed beamformer may be updated from time to time (=> adaptive). In an embodiment, an own-speech beamformer focused on the user's mouth and an ambient sound beamformer focused on sound sources of interest in the user's environment are created simultaneously using the electrical input signal.

[0051] The adaptively updated beamformer weights, for example, the frequency-dependent adaptive factor β(k) may be the optimal adaptive factor β obtained from the electrical input signal. mic (k) (See, for example, Figure 2 The lower part) and the adaptive factor β obtained from the feedback estimator FBE (k) (See, for example, Figure 4 The synthetic adaptive factor βmix (k) can be the optimal adaptive factor β mic (k) and the adaptive factor β based on the feedback estimator FBE Linear combination of (k):

[0052] β(k)=α·β mic (k)+(1-α)·β FBE (k)

[0053] wherein α is a (e.g. real number) weighting factor having a value between 0 and 1. The weighting factor α may be fixed or adaptively determined. The weighting factor α may, for example, be determined depending on the input level (e.g. level L of the electrical input signal). The weighting factor α may, for example, increase from 0 to 1 as the level (L) increases, e.g. in a stepped or piecewise linear or monotonic (e.g. S-shaped or quasi-S-shaped) manner. Values ​​of the weighting factor α close to 0 represent a configuration or an acoustic situation that focuses on reducing external noise in the (far-field) acoustic input signal. Values ​​of the weighting factor α close to 1 represent a configuration or an acoustic situation that focuses on reducing feedback from the (near-field) acoustic input signal (loudspeaker of the hearing device).

[0054] The hearing device may comprise a detector of the current acoustic environment, which detector provides an environment detection signal indicative of the current feedback situation.

[0055] The hearing device may be configured to apply the associated set of predetermined feedback estimates to provide the second set of beamformers C F1 ,C F2 .

[0056] The hearing device may comprise a feedback suppression system for suppressing feedback from said output transducer to at least one of said input transducers. The hearing device may comprise a feedback suppression system for suppressing feedback from said output transducer to each of said plurality of input transducers. The feedback suppression system may for example be configured to subtract a current estimate of a current feedback path from said output transducer to each of said input transducers from the respective electrical input signal (or a signal derived therefrom). The feedback system may comprise a respective subtraction unit for subtracting an estimate of a current feedback path for a given input transducer from the electrical input signal provided by the input transducer. In an embodiment, the estimate of the current feedback path is provided in the time domain. In an embodiment, the estimate of the current feedback path is provided in the (time-) frequency domain. The feedback suppression system may for example be configured to estimate the feedback paths of all M input transducers and to subtract the current estimate of the feedback paths from the respective (current) electrical input signal (or a processed version thereof), see for example Figure 4 An additional set of analysis filter banks may be used to transform the estimated time-domain feedback path estimator into a time-frequency domain feedback estimator.

[0057] The hearing device may consist of or include a hearing aid, a headset, an ear protection device or a combination thereof.It should be noted that in a headset, the target sound will typically be the wearer's own voice of the headset.

[0058] The hearing device may comprise an ITE part adapted to be located at or in the ear canal of a user, the ITE part comprising a housing comprising a seal towards the wall or ear canal such that the ITE part fits closely to the ear canal wall or at least provides a controlled or minimal sound leakage path, the ITE part comprising at least two microphones located outside the seal, facing the environment and at least one microphone located inside the seal and facing the ear drum. The microphone inside the seal mainly records feedback signals, for which reason it no longer introduces noise, which has been removed by the beamforming signals obtained from the two microphones outside the seal.

[0059] First Another Hearing Device

[0060] In one aspect, the present invention provides a first further hearing device. The hearing device, such as a hearing aid, is configured to be located at or in an ear of a user. The hearing device comprises an ITE part adapted to be located at or in an ear canal of a user. The ITE part comprises:

[0061] - a housing configured to be located at least partially in the ear canal of a user, the housing possibly comprising a seal towards the wall or ear canal so that the ITE portion fits closely against the ear canal wall or at least provides a controlled or minimal sound leakage path;

[0062] at least three input transducers for providing respective electrical input signals, wherein when the ITE part is mounted at or in the ear canal in operation, at least two input transducers face the environment and provide respective electrical input signals representative of sounds in the user's environment, and at least one input transducer faces the ear drum and provides at least one electrical input signal representative of sounds reflected from the ear drum;

[0063] - an output transducer for providing a stimulus perceivable as sound by a user based on said electrical input signal or a processed version thereof;

[0064] - a beamformer filter unit connected to the at least three input transducers and the output transducer, configured to provide a spatially filtered signal based on at least three electrical input signals and appropriate beamformer weights;

[0065] - wherein the beamformer filter unit comprises

[0066] a first beamformer for spatially filtering sounds in the environment based on electrical input signals from the at least two input transducers towards the environment; and

[0067] a second beamformer for spatially filtering the sound reflected from the eardrum based on at least one of the at least one electrical input signal from the at least one input transducer towards the eardrum and the electrical input signals from the at least two input transducers towards the environment.

[0068] The hearing device features outlined above for the hearing device and the hearing device features outlined below under the heading "Additional hearing aid features" may be combined with the first further hearing device, where appropriate.

[0069] The microphone inside the seal mainly records the feedback signal, for which reason it no longer introduces noise, which has been removed by the beamformed signals obtained from the two microphones outside the seal.

[0070] The first and second beamformers are preferably available simultaneously.

[0071] When the ITE part is mounted in the ear canal in operation, the stimulation may be directed to the ear drum.The output transducer may be a loudspeaker.

[0072] At least two microphones facing the environment and at least one input transducer facing the ear drum are located on each side of the seal.

[0073] Directional weights for different channels can be used for different purposes. In channels where feedback is dominant, the directional system can be used for feedback cancellation, while in channels where feedback is not significant, the directional system can be used for noise reduction (of external noise sources or microphone noise).

[0074] Second another hearing device

[0075] In one aspect, the present invention provides a second hearing device. The hearing device, such as a hearing aid, is configured to be located at or in an ear of a user. The hearing device comprises

[0076] - at least two input transducers for providing respective electrical input signals;

[0077] - an output transducer for providing a stimulus perceivable as sound by a user based on said electrical input signal or a processed version thereof;

[0078] - a feedback estimation unit for providing a feedback estimate of a current feedback path from the output converter to at least one of the at least two input converters;

[0079] - a beamformer filtering unit connected to said at least two input transducers and said output transducer, configured to provide a spatially filtered signal based on said at least two electrical input signals and appropriate beamformer weights;

[0080] - a post filter connected to the beamformer filtering unit, configured to provide a frequency and time dependent gain to be applied to the spatially filtered signal so as to further reduce noise in the spatially filtered signal;

[0081] - wherein said beamformer filter unit and / or said post filter are updated using said feedback estimate.

[0082] The hearing device features outlined above for the hearing device and the first further hearing device and the hearing device features outlined below under the heading "Further hearing aid features" (and detailed in the detailed description and defined in the claims) may be combined with the second further hearing device, where appropriate.

[0083] The portion of the beamformer filter unit that provides the spatially filtered signal may be updated using the feedback estimate.

[0084] The postfilter may determine a gain based on a noise estimate provided by the feedback estimate.

[0085] A beamformer filter unit providing a spatially filtered signal and a postfilter providing a frequency and time dependent gain to be applied to the spatially filtered signal may be updated based on the feedback estimate.

[0086] The hearing device may be configured to provide a feedback estimate for each of the at least two input transducers.The beamformer filter unit and / or the postfilter may be updated using each of the individual feedback estimates or a combination of the feedback estimates, such as an average or maximum value.

[0087] Hearing Device Features

[0088] Where appropriate, the following features may be combined with the hearing device described above (and detailed in the detailed description and defined in the claims) and the first and second further hearing devices.

[0089] In an embodiment, the hearing device is adapted to provide a frequency dependent gain and / or a level dependent compression and / or a frequency transposition of one or more frequency ranges to one or more other frequency ranges (with or without frequency compression) to compensate for a hearing impairment of the user. In an embodiment, the hearing device comprises a signal processor for enhancing an input signal and providing a processed output signal.

[0090] The hearing device comprises an output unit for providing a stimulus perceived by a user as an acoustic signal based on the processed electrical signal. In an embodiment, the output unit comprises an output transducer. In an embodiment, the output transducer comprises a receiver (speaker) for providing the stimulus as an acoustic signal to the user. In an embodiment, the output transducer comprises a vibrator for providing the stimulus as a mechanical vibration of a skull to the user (e.g. in a bone-attached or bone-anchored or bone-conducting hearing device).

[0091] In an embodiment, the hearing device comprises a further output unit for providing stimulation to another user, e.g. as a remote input for a telephone conversation. These output units may be connected to a signal processor so as to enable control of the output signals presented via the respective output units (e.g. a transmitter or another output transducer), different signals presented via different output units, e.g. one signal intended for presentation to the user and another signal intended for presentation to an external device (e.g. another person). The hearing device may be configured, e.g. in a specific operating mode (e.g. a communication or telephone mode), to pick up the user's own voice (e.g. via a predetermined (or adaptive) beamformer focused on the user's mouth).

[0092] The hearing device comprises an input unit for providing an electrical input signal representing a sound. In an embodiment, the input unit comprises an input transducer, such as a microphone, for converting the input sound into the electrical input signal. In an embodiment, the input unit comprises a wireless receiver for receiving a wireless signal comprising the sound and providing the electrical input signal representing the sound. The number of input transducers, such as microphones, may be greater than or equal to 2, such as greater than or equal to 3, such as greater than or equal to 4.

[0093] A hearing device comprises a directional microphone system adapted to spatially filter sounds from the environment so as to enhance a target sound source among a plurality of sound sources in the local environment of a user wearing the hearing device. In an embodiment, the directional system is adapted to detect (e.g. adaptively detect) from which direction a particular portion of a microphone signal (e.g. a target signal and / or a noise signal) originates. This can be achieved in a number of different ways, such as described in the prior art. In hearing devices, microphone array beamformers are often used to spatially attenuate background noise sources. Many beamformer variants can be found in the literature. Minimum Variance Distortionless Response (MVDR) beamformers are widely used in microphone array signal processing. Ideally, an MVDR beamformer keeps the signal from the target direction (also called the line-of-sight direction) unchanged, while maximally attenuating sound signals from other directions. A Generalized Sidelobe Canceller (GSC) structure is an equivalent representation of an MVDR beamformer, which provides computational and digital representation advantages over a direct implementation in its original form.

[0094] In an embodiment, the hearing device comprises an antenna and a transceiver circuit (e.g. a wireless receiver) for receiving a direct electric input signal from another device, such as from an entertainment device (e.g. a television), a communication device, a wireless microphone or another hearing device. In an embodiment, the direct electric input signal represents or comprises an audio signal and / or a control signal and / or an information signal.

[0095] Preferably, the frequency used for establishing the communication link between the hearing device and the other device is below 70 GHz, e.g. in the range from 50 MHz to 70 GHz, e.g. above 300 MHz, e.g. in the ISM range above 300 MHz, e.g. in the 900 MHz range or in the 2.4 GHz range or in the 5.8 GHz range or in the 60 GHz range (ISM = Industrial, Scientific and Medical, such standardized ranges are defined, for example, by the International Telecommunication Union ITU). In an embodiment, the wireless link is based on a standardized or proprietary technology. In an embodiment, the wireless link is based on Bluetooth technology (e.g. Bluetooth low power technology).

[0096] In an embodiment, the hearing device is a portable device, such as a device comprising a local energy source, such as a battery, eg a rechargeable battery.

[0097] In an embodiment, the hearing device comprises a forward or signal path between an input unit, such as an input transducer, e.g. a microphone or a microphone system and / or a direct electrical input, such as a wireless receiver, and an output unit, such as an output transducer. In an embodiment, a signal processor is located in the forward path. In an embodiment, the signal processor is adapted to provide a frequency dependent gain according to a specific need of a user. In an embodiment, the hearing device comprises an analysis path with functional parts for analyzing the input signal, such as determining the level, modulation, signal type, an estimate of acoustic feedback, etc. In an embodiment, part or all of the signal processing of the analysis path and / or the signal path is performed in the frequency domain. In an embodiment, part or all of the signal processing of the analysis path and / or the signal path is performed in the time domain.

[0098] In an embodiment, the hearing device, such as a microphone unit and / or a transceiver unit, comprises a TF conversion unit for providing a time-frequency representation of an input signal. In an embodiment, the time-frequency representation comprises an array or a mapping of respective complex or real values ​​of the signal in question in a certain time and frequency range. In an embodiment, the TF conversion unit comprises a filter bank for filtering the (time-varying) input signal and providing a plurality of (time-varying) output signals, each output signal comprising a distinct frequency range of the input signal. In an embodiment, the TF conversion unit comprises a Fourier transform unit for converting the time-varying input signal into a (time-varying) signal in the (time-)frequency domain. In an embodiment, the hearing device considers the time-frequency representation from the minimum frequency f min To the maximum frequency f maxThe frequency range of includes a portion of the typical human hearing range from 20 Hz to 20 kHz, for example a portion of the range from 20 Hz to 12 kHz. Typically, the sampling rate f s Greater than or equal to the maximum frequency f max twice, that is, f s ≥2f max In an embodiment, the signal of the forward path and / or analysis path of the hearing device is split into NI frequency bands (e.g. of uniform width), wherein NI is for example greater than 5, such as greater than 10, such as greater than 50, such as greater than 100, such as greater than 500, at least parts of which are processed separately. In an embodiment, the hearing aid is adapted to process the signal of the forward and / or analysis path at NP different frequency channels (NP≤NI). The frequency channels may be of uniform or non-uniform width (e.g. the width increases with frequency), overlapping or non-overlapping.

[0099] In an embodiment, the hearing device comprises a plurality of detectors configured to provide status signals related to a current network environment (e.g. a current acoustic environment) of the hearing device, and / or to a current state of a user wearing the hearing device, and / or to a current state or operating mode of the hearing device. Alternatively or additionally, one or more of the detectors may form part of an external device that communicates with the hearing device (e.g. wirelessly). The external device may, for example, comprise another hearing device, a remote control, an audio transmission device, a phone (e.g. a smartphone), an external sensor, etc.

[0100] In an embodiment, one or more of the plurality of detectors operates on full-band signals (time domain).In an embodiment, one or more of the plurality of detectors operates on band-split signals ((time-)frequency domain), e.g. in a limited plurality of frequency bands.

[0101] In an embodiment, a plurality of detectors include a level detector for estimating the current level of the signal of the forward path. In an embodiment, the predetermined criterion includes whether the current level of the signal of the forward path is above or below a given (L-) threshold. In an embodiment, the level detector acts on a full-band signal (time domain). In an embodiment, the level detector acts on a band split signal ((time-) frequency domain).

[0102] In a particular embodiment, the hearing device comprises a voice detector (VD) for estimating whether (or with what probability) an input signal (at a particular point in time) comprises a voice signal. In the present specification, a voice signal comprises a speech signal from a human being. It may also comprise other forms of vocalizations (such as singing) produced by a human speech system. In an embodiment, the voice detector unit is adapted to classify the user's current acoustic environment as a "voice" or a "no-voice" environment. This has the advantage that time periods comprising an electric microphone signal of human vocalizations (such as speech) in the user's environment can be identified and thus separated from time periods comprising only (or mainly) other sound sources (such as artificially produced noise). In an embodiment, the voice detector is adapted to detect the user's own voice as "voice" as well. As an alternative, the voice detector is adapted to exclude the user's own voice from the detection of "voice".

[0103] In an embodiment, the hearing device comprises an own voice detector for estimating whether (or with what probability) a particular input sound (e.g. voice, such as speech) originates from the voice of a user of the system. In an embodiment, the microphone system of the hearing device is adapted to be able to distinguish between the user's own voice and the voice of another person and possibly from unvoiced sounds.

[0104] In an embodiment, the plurality of detectors comprises a motion detector, such as an accelerometer. In an embodiment, the motion detector is configured to detect movement of the user's facial muscles and / or bones, such as caused by speech or chewing (such as jaw movement) and provide a detector signal indicating the movement.

[0105] In connection with removing feedback, own voice or jaw movement may change the feedback path. Therefore, when own voice or jaw movement has been detected, it may be beneficial to increase the adaptation rate.

[0106] In an embodiment, the hearing device comprises a classification unit configured to classify the current situation based on the input signal from (at least part of) the detector and possibly other inputs. In this specification, "current situation" is defined by one or more of the following:

[0107] a) the physical environment (e.g. including the current electromagnetic environment, such as the presence of electromagnetic signals (including audio and / or control signals) intended or not intended to be received by the hearing device, or other properties of the current environment other than acoustics);

[0108] b) Current acoustic conditions (input level, feedback, etc.);

[0109] c) the user’s current mode or state (motion, temperature, cognitive load, etc.);

[0110] d) The current mode or status of the hearing device and / or another device communicating with the hearing device (selected program, time elapsed since last user interaction, etc.).

[0111] In an embodiment, the hearing device comprises an acoustic (and / or mechanical) feedback suppression system.

[0112] In an embodiment, the hearing device comprises a feedback estimation unit for providing a feedback signal representing an estimate of an acoustic feedback path and a combining unit, such as a subtraction unit, for subtracting the feedback signal from a signal of a forward path (e.g. picked up by an input transducer of the hearing device). In an embodiment, the feedback estimation unit comprises an updating part comprising an adaptive algorithm and a variable filter part for filtering the input signal according to variable filter coefficients determined by the adaptive algorithm, wherein the updating part is configured to update the input signal at a configurable updating frequency f upd The filter coefficients of the variable filter part are updated. In an embodiment, the hearing device is configured such that the configurable update frequency f upd With the maximum value f upd,max In an embodiment, the maximum value f upd,max is the sampling frequency f of the AD converter of the hearing device s Part of upd,max =f s / D).

[0113] The updating part of the adaptive filter comprises an adaptive algorithm for calculating updated filter coefficients for transmission to the variable filter part of the adaptive filter. The timing of the calculation of the updated filter coefficients and / or the transmission from the updating part to the variable filter part may be controlled by the activation control unit. The timing of the updating (e.g. its specific time point, and / or its updating frequency) may preferably be influenced by a variety of different properties of the signal of the forward path. The updating control scheme is preferably supported by one or more detectors of the hearing device, preferably included in a predetermined criterion including the detector signal.

[0114] In an embodiment, the hearing device further comprises other suitable functions for the application in question, such as compression, noise reduction, active noise cancellation, etc.

[0115] In an embodiment, the hearing device comprises a listening device such as a hearing aid, a hearing instrument such as a hearing instrument adapted to be located at the ear of a user or fully or partially in the ear canal, such as headphones, earphones, ear protection or a combination thereof.

[0116] application

[0117] In one aspect, there is provided an application of a hearing device as described above, in detail in the "Detailed Description" section and in the claims. In an embodiment, there is provided an application in a system comprising audio distribution, such as a system comprising a microphone and a loudspeaker that are sufficiently close to each other to cause feedback from the loudspeaker to the microphone during user operation. In an embodiment, there is provided an application in a system comprising one or more hearing aids (such as hearing instruments), headphones, earphones, active ear protection systems, etc., such as use in hands-free telephone systems, teleconferencing systems, broadcast systems, karaoke systems, classroom amplification systems, etc.

[0118] method

[0119] In one aspect, the present invention provides a method of suppressing feedback in a hearing device, the hearing device being adapted to be located at or in an ear of a user or to be fully or partially implanted in a head at the ear, the hearing device comprising a plurality of input transducers connected to each other and an output transducer. The method comprises:

[0120] - providing a plurality of electrical input signals representative of sounds in the user's environment;

[0121] - providing a stimulus perceivable as sound by a user based on the electrical input signal or a processed version thereof;

[0122] - providing a spatially filtered signal based on the plurality of electrical input signals and adaptively updated beamformer weights;

[0123] - providing a feedback estimate of the current feedback path from said output transducer to each of said input transducers; and

[0124] - causing at least one of said adaptively updated beamformer weights to be updated in dependence on a feedback path estimate.

[0125] When appropriately replaced by corresponding processes, part or all of the structural features of the apparatus described above, described in detail in the "Detailed Description" or defined in the claims can be combined with the implementation of the method of the present invention, and vice versa. The implementation of the method has the same advantages as the corresponding apparatus.

[0126] The method may comprise providing three or more electrical input signals, wherein at least some of them are used for spatial filtering and reduction of noise in said sound in the environment, and wherein at least some of them are used for feedback cancellation, and wherein at least one electrical input signal is used for both.

[0127] Directional weights for different channels can be used for different purposes. In channels where feedback is dominant, the directional system can be used for feedback cancellation, while in channels where feedback is not significant, the directional system can be used for noise reduction (of external noise sources or microphone noise).

[0128] Computer readable medium

[0129] The present invention further provides a tangible computer-readable medium storing a computer program including program code, which, when the computer program is run on a data processing system, causes the data processing system to perform at least part (such as most or all) of the steps of the method described above, described in detail in the "Specific Implementation Methods" and defined in the claims.

[0130] As an example and not limitation, the aforementioned tangible computer readable medium may include RAM, ROM, EEPROM, CD-ROM or other optical disk storage, magnetic disk storage or other magnetic storage device, or any other medium that can be used to execute or store the desired program code in the form of instructions or data structures and can be accessed by a computer. As used herein, disks include compact disks (CDs), laser disks, optical disks, digital versatile disks (DVDs), floppy disks, and Blu-ray disks, wherein these disks usually reproduce data magnetically, while these disks can reproduce data optically with lasers. Combinations of the above disks should also be included in the scope of computer readable media. In addition to being stored on tangible media, computer programs can also be transmitted via transmission media such as wired or wireless links or networks such as the Internet and loaded into a data processing system to be run at a location different from the tangible media.

[0131] Computer Programs

[0132] In addition, the present application provides a computer program (product) comprising instructions, which, when executed by a computer, causes the computer to execute (the steps of) the method described above, described in detail in the “Detailed Description” and defined in the claims.

[0133] Data processing system

[0134] On the one hand, the present invention further provides a data processing system comprising a processor and a program code, wherein the program code enables the processor to perform at least part (such as most or all) of the steps of the method described above, described in detail in the "Specific Implementation Method" and defined in the claims.

[0135] Hearing System

[0136] In another aspect, a hearing system comprises a hearing device as described above, in detail in the “Detailed Description of the Invention” and in the claims, and further an auxiliary device is provided.

[0137] In an embodiment, the hearing system is adapted to establish a communication link between the hearing device and the auxiliary device such that information (such as control and status signals, possibly audio signals) can be exchanged or forwarded from one device to the other.

[0138] In an embodiment, the hearing system includes an auxiliary device, such as a remote control, a smart phone, or other portable or wearable electronic device such as a smart watch, etc. The hearing system may also include a device (such as a microphone or other sensor or processing device) located elsewhere on the user's body (such as at the other ear) or a device worn by or located at another person.

[0139] In an embodiment, the auxiliary device is or comprises a remote control for controlling the functions and operation of the hearing device. In an embodiment, the functions of the remote control are implemented in a smartphone, which may run an APP enabling control of the functions of the audio processing device via the smartphone (the hearing device comprises a suitable wireless interface to the smartphone, e.g. based on Bluetooth or some other standardized or proprietary solution).

[0140] In an embodiment, the auxiliary device is or comprises an audio gateway device which is adapted to receive a plurality of audio signals (e.g. from an entertainment device such as a TV or a music player, from a telephone device such as a mobile phone or from a computer such as a PC) and to select and / or combine an appropriate signal (or combination of signals) of the received audio signals for transmission to the hearing device.

[0141] In an embodiment, the auxiliary device is or comprises another hearing device.

[0142] In an embodiment, the hearing system comprises two hearing devices adapted to implement a binaural hearing system, such as a binaural hearing aid system.

[0143] APP

[0144] On the other hand, the present invention also provides a non-transient application called APP. The APP includes executable instructions configured to run on an auxiliary device to implement a user interface for the hearing device or hearing system described above, described in detail in the "Detailed Description of the Invention" and defined in the claims. In an embodiment, the APP is configured to run on a mobile phone such as a smart phone or another portable device that enables communication with the hearing device or hearing system.

[0145] definition

[0146] In this specification, a "hearing device" refers to a device such as a hearing aid, for example a hearing instrument or an active ear protection device or other audio processing device, which is suitable for improving, enhancing and / or protecting the hearing ability of a user by receiving an acoustic signal from the user's environment, generating a corresponding audio signal, possibly modifying the audio signal, and providing the possibly modified audio signal as an audible signal to at least one ear of the user. A "hearing device" also refers to a device such as a headset or an earphone, which is suitable for electronically receiving an audio signal, possibly modifying the audio signal, and providing the possibly modified audio signal as an audible signal to at least one ear of the user. The audible signal can be provided, for example, in the form of an acoustic signal radiated into the outer ear of the user, an acoustic signal transmitted to the inner ear of the user as a mechanical vibration through the bone structure of the user's head and / or through parts of the middle ear, and an electrical signal transmitted directly or indirectly to the cochlear nerve of the user.

[0147] The hearing device may be configured to be worn in any known manner, such as as a unit worn behind the ear (with a tube directing the radiated acoustic signal into the ear canal or with an output transducer such as a loudspeaker arranged close to or in the ear canal), as a unit arranged wholly or partly in the pinna and / or ear canal, as a unit connected to a fixed structure implanted in the skull such as a vibrator, or as a connectable or wholly or partly implanted unit, etc. The hearing device may comprise a single unit or several units in electronic communication with each other. The loudspeaker may be arranged in a housing together with the other components of the hearing device, or it may be an external unit itself (possibly in combination with a flexible guiding element such as a dome-shaped element).

[0148] More generally, a hearing device comprises an input transducer for receiving an acoustic signal from the user's environment and providing a corresponding input audio signal and / or a receiver for receiving the input audio signal electronically (i.e. wired or wirelessly), a (typically configurable) signal processing circuit (e.g. a signal processor, e.g. comprising a configurable (programmable) processor, e.g. a digital signal processor) for processing the input audio signal, and an output unit for providing an audible signal to the user based on the processed audio signal. The signal processor may be adapted to process the input signal in the time domain or in a plurality of frequency bands. In some hearing devices, an amplifier and / or a compressor may constitute the signal processing circuit. The signal processing circuit typically comprises one or more (integrated or separate) memory elements for executing a program and / or for storing parameters used (or likely to be used) in the processing and / or for storing information suitable for the functioning of the hearing device and / or for storing information (e.g. processed information, e.g. provided by the signal processing circuit) for use, e.g. in conjunction with an interface to a user and / or to an interface to a programming device. In some hearing devices, the output unit may comprise an output transducer, e.g. a loudspeaker for providing an air-borne acoustic signal or a vibrator for providing a structure- or liquid-borne acoustic signal. In some hearing devices, the output unit may include one or more output electrodes for providing an electrical signal (eg a multi-electrode array for electrical stimulation of the cochlear nerve).

[0149] In some hearing devices, the vibrator may be adapted to transmit structure-borne acoustic signals to the skull percutaneously or by the skin. In some hearing devices, the vibrator may be implanted in the middle ear and / or the inner ear. In some hearing devices, the vibrator may be adapted to provide structure-borne acoustic signals to the bones of the middle ear and / or the cochlea. In some hearing devices, the vibrator may be adapted to provide liquid-borne acoustic signals to the cochlear fluid, for example, through the oval window. In some hearing devices, the output electrode may be implanted in the cochlea or on the inside of the skull and may be adapted to provide electrical signals to the hair cells of the cochlea, one or more auditory nerves, the auditory brainstem, the auditory midbrain, the auditory cortex, and / or other parts of the cerebral cortex.

[0150] A hearing device such as a hearing aid may be adapted to the needs of a specific user such as hearing loss. The configurable signal processing circuit of the hearing device may be adapted to apply frequency- and level-dependent compression amplification of an input signal. The customized frequency- and level-dependent gain (amplification or compression) may be determined during the fitting process by the fitting system based on the user's hearing data such as an audiogram using the basic principles of fitting (e.g. adaptation to speech). The frequency- and level-dependent gain may, for example, be embodied in a processing parameter, uploaded to the hearing device, for example via an interface to a programming device (fitting system), and used by a processing algorithm executed by the configurable signal processing circuit of the hearing device.

[0151] "Hearing system" refers to a system including one or two hearing devices. "Binaural hearing system" refers to a system including two hearing devices and adapted to provide audible signals to the two ears of a user in a coordinated manner. A hearing system or binaural hearing system may also include one or more "auxiliary devices" that communicate with the hearing device and affect and / or benefit from the function of the hearing device. An auxiliary device may be, for example, a remote controller, an audio gateway device, a mobile phone (such as a smart phone) or a music player. A hearing device, a hearing system or a binaural hearing system may be used, for example, to compensate for the hearing loss of a hearing-impaired person, enhance or protect the hearing ability of a normal hearing person, and / or transmit an electronic audio signal to a person. A hearing device or a hearing system may, for example, form part of a broadcasting system, an active ear protection system, a hands-free telephone system, a car audio system, an entertainment (such as karaoke) system, a teleconferencing system, a classroom amplification system, or interact with it.

[0152] Embodiments of the present invention may be used, for example, in the applications mentioned in this application. BRIEF DESCRIPTION OF THE DRAWINGS

[0153] Various aspects of the present invention will be best understood from the detailed description below in conjunction with the accompanying drawings. For clarity, the drawings are schematic and simplified, and only the details necessary for understanding the present invention are given, while other details are omitted. Throughout the specification, the same reference numerals are used for the same or corresponding parts. The various features of each aspect may be combined with any or all features of the other aspects. These and other aspects, features and / or technical effects will be apparent from and illustrated in conjunction with the following figures, in which:

[0154] Figure 1A and 1B A hearing device comprising two microphones located in the ear canal and adapted to cancel the sound propagated in the feedback path by applying a fixed or adaptive directional gain is shown.

[0155] Figure 2 An embodiment of a two-microphone MVDR beamformer according to the present invention is shown.

[0156] Figure 3 A hearing device comprising a beamformer filter unit according to the invention is shown, wherein the beamformer filter unit provides a target cancellation beamformer for cancelling sound from a target signal in the acoustic far field as shown by a cardioid.

[0157] Figure 4 It shows that Figure 2 Another embodiment of a dual-microphone MVDR beamformer is shown in .

[0158] Figure 5An embodiment of a RITE type hearing device according to the invention is schematically shown, comprising a BTE part, an ITE part and a connecting element.

[0159] Figure 6 A schematic block diagram of an embodiment of a hearing device comprising two microphones according to the invention is shown.

[0160] Fig. 7A An embodiment of a hearing device according to the invention comprising two microphones located in the ITE part is shown.

[0161] Figure 7B It shows that Fig. 7A A schematic block diagram of an embodiment of a hearing device is shown in .

[0162] Figure 7C An embodiment of a hearing device according to the invention comprising three microphones in the ITE part is shown.

[0163] Fig.7D It shows that Figure 7C A schematic block diagram of an embodiment of a hearing device is shown in .

[0164] Fig. 7E An embodiment of a hearing device according to the invention is shown comprising four microphones, wherein two microphones are located in the BTE part and two microphones are located in the ITE part.

[0165] Figure 7F It shows that Fig. 7E A schematic block diagram of an embodiment of a hearing device is shown in .

[0166] Fig. 8A An embodiment of a hearing device according to the invention comprising three microphones in the ITE part is shown.

[0167] Figure 8B It shows that Fig. 8A A schematic block diagram of an embodiment of a hearing device is shown in .

[0168] Fig.9A A first embodiment of a hearing device is shown comprising two input transducers (eg microphones) for cancelling noise in the environment and feedback from the output transducers (eg loudspeakers) to the input transducers (microphones).

[0169] Fig. 9B A second embodiment of a hearing device is shown comprising two input transducers for cancelling noise in the environment and feedback from the output transducer to the input transducer.

[0170] Fig. 9CA third embodiment of a hearing device is shown comprising two input transducers for cancelling noise in the environment and feedback from the output transducer to the input transducer.

[0171] By the detailed description given below, the further scope of application of the present invention will be apparent. However, it should be understood that while the detailed description and specific examples show the preferred embodiments of the present invention, they are only provided for illustrative purposes. For those skilled in the art, based on the following detailed description, other embodiments of the present invention will be apparent. DETAILED DESCRIPTION

[0172] The detailed description proposed below in conjunction with the accompanying drawings serves as a description of a variety of different configurations. The detailed description includes specific details for providing a thorough understanding of a number of different concepts. However, it is apparent to those skilled in the art that these concepts can be implemented without these specific details. Several aspects of the apparatus and method are described by a number of different blocks, functional units, modules, components, circuits, steps, processes, algorithms, etc. (collectively referred to as "elements"). Depending on the specific application, design limitations or other reasons, these elements can be implemented using electronic hardware, computer programs or any combination thereof.

[0173] The electronic hardware may include microprocessors, microcontrollers, digital signal processors (DSPs), field programmable gate arrays (FPGAs), programmable logic devices (PLDs), gating logic, discrete hardware circuits, and other appropriate hardware configured to perform a number of different functions described in this specification. Computer programs should be broadly interpreted as instructions, instruction sets, codes, code segments, program codes, programs, subroutines, software modules, applications, software applications, software packages, routines, subroutines, objects, executables, execution threads, programs, functions, etc., whether referred to as software, firmware, middleware, microcode, hardware description language, or other names.

[0174] The present application relates to the field of hearing devices such as hearing aids, and in particular to feedback from an output transducer to an input transducer of a hearing device.

[0175] The present invention proposes an anti-feedback solution based on the spatial resolution of the signal.

[0176] Feedback in hearing aids is usually reduced by subtracting the estimated feedback path from the microphone signal. Usually, hearing aids contain more than one microphone. Thereby, the spatial information of the microphone can be used to cancel the feedback. On the one hand, we consider special microphone configurations (see Figure 1A , 1B ), which is well suited for directional feedback cancellation without changing the target signal.

[0177] Figure 1A and 1BA hearing device comprising two microphones located in the ear canal adapted to cancel the sound propagated in the feedback path by applying a fixed or adaptive directional gain is shown.

[0178] Adaptive beamforming in hearing instruments aims to cancel unwanted noise under the constraint that the sound from the target direction is not altered. Examples of such adaptive systems are Figure 2 As shown in FIG, the output signal Y(k) in the kth channel is based on a linear combination of two fixed beamformers C1(k) and C2(k), i.e., Y(k)=C1(k)-β(k)C2(k), where C1(k) and C2(k) are preferably orthogonal beamformers, and while C1(k) maintains the target direction, C2(k) is a beamformer that cancels the sound from the target direction.

[0179] Figure 2 An embodiment of a two-microphone MVDR beamformer according to the present invention is shown. Based on two microphones, two fixed beamformers are created: a beamformer C1 that does not change the signal from the target direction and a (orthogonal) beamformer C2 that cancels the signal from the target direction. The resulting directional signal Y(k) = C1(k) - β(k)C2(k), where

[0180]

[0181] The noise is minimized under the constraint that the signal from the target direction is not changed. LP refers to averaging of the signal, for example by a first-order IIR low-pass filter.

[0182] The adaptive factor β(k) is the weight applied to the target cancellation beamformer. Thus, knowing that the target direction is not changed, we can adjust β(k). In the case where we want to cancel feedback, all external sounds are considered as sounds of interest. With the selected microphone configuration, all external sounds will pass through the first microphone before reaching the second microphone, e.g. Figure 3 as shown in .

[0183] Figure 3A hearing device comprising a beamformer filter unit according to the invention is shown, wherein the beamformer filter unit provides a target cancellation beamformer for cancelling sounds from a target signal in the acoustic far field as shown by a cardioid. The cardioid is shown here as a directivity diagram, but in reality the beam pattern depends not only on the sound source direction, it also varies as a function of the distance between the sound source and the microphone. The target cancellation beamformer is configured to cancel signals incident on the hearing aid. Due to the microphone configuration, external sounds first have to pass through the first microphone and secondly have to pass through the second microphone. Most external sounds will thus have approximately the same delay as seen by the hearing aid. Thereby, the target cancellation beamformer will work efficiently for most target directions.

[0184] Another difference between external sounds and feedback sounds is that feedback sounds are most likely to have the highest sound pressure level at the inner microphone, whereas external sounds are most likely to have the highest sound pressure level at the outer microphone. In an embodiment, the hearing device is configured to compare the levels of the inner and outer microphones at a given point in time, e.g. when feedback is detected.

[0185] In other words, all external sounds may be considered (from the hearing instrument microphone) as sounds coming from one distinct direction. We therefore propose to estimate the target cancellation beamformer such that it minimizes sounds incident from all external directions. This may be achieved, for example, based on impulse response recordings of external sounds from a number of different external directions (e.g. with predetermined weights determined based on the measurements). As an alternative, the target cancellation beamformer may be estimated based on responses from a preferred direction (i.e. a direction is selected and a fixed beamformer (e.g. beamformer weights) is determined for this direction, preferably the front direction or the self-voice direction). A third option is to adapt the target cancellation beamformer to the current listening direction, i.e. to cancel external sounds at any time. Such an adaptive target cancellation beamformer may be updated when the external sound is much louder than the feedback signal. The task of the target cancellation BF is to estimate the "noise", which is the feedback "from the ear drum". Due to compression, we have relatively less feedback at high external input levels than at low input levels, because we generally need less amplification at high input levels.

[0186] Compared to the typical update of the adaptation coefficient β(k), which is directly based on the microphone signal, we propose to update this coefficient based on the feedback path estimate.

[0187] An advantage is that the adaptive beamformer will thereby be less dependent on external sounds. A disadvantage may be that the beamformer is dependent on the feedback path estimate and, for this reason, cannot react faster than the feedback path estimate. Still, it is possible that the adaptive beamformer will be able to attenuate the feedback path estimate even if the beam pattern is not perfectly adjusted.

[0188] Some feedback path estimates are more reliable than others. Thus, not all values ​​of β(k) will represent possible feedback. Considering adaptation, the value β(k) may thus provide an estimate of how reliable the current (single-microphone) feedback path estimate is.

[0189] Figure 4 Shown as Figure 2 Another embodiment of a two-microphone MVDR beamformer is shown in . The beamformer filter unit is based on two fixed beamformers: a beamformer C1 that does not change the signal from the target direction, and a (orthogonal) beamformer C2 that cancels the signal from the target direction. The target direction is the direction of all external sounds, which can be seen as a single direction due to the microphone configuration. The resulting directional signal is still given by Y(k) = C1(k) - β(k)C2(k), but with Figure 2 Instead, the adaptive factor β(k) is estimated based on another set of fixed beamformers with the same weights (w 11 ,w 21 ,w 12 ,w 22 ) but in this case is used as input to the (frequency domain) feedback path estimator The adaptation factor is then given by:

[0190]

[0191] The advantage of using the feedback path estimate compared to the microphone signal is that the updating of the adaptive beam pattern will be less affected by external sounds.

[0192] Figure 5 An embodiment of a hearing device according to the invention is schematically shown. The hearing device HD, such as a hearing aid, is of a specific type (sometimes referred to as a receiver-in-the-ear or RITE type) and comprises a BTE part (BTE) adapted to be located at or behind the ear of a user and an ITE part (ITE) adapted to be located in or at the ear canal of a user and comprising a receiver (speaker). The BTE part and the ITE part are connected (e.g. electrically connected) via a connection element IC and internal wiring in the ITE and BTE parts (e.g. see wiring Wx in the BTE part).

[0193] exist Figure 5 In an embodiment of the hearing device, the BTE part comprises two input units (M BTE1 ,M BTE2 , see also e.g. Figure 2 , 3 , M2, M2 in 4), which includes a corresponding input transducer (such as a microphone) (M BTE1 ,M BTE2 ), each input transducer is used to provide a signal representing an input sound signal (SBTE ) (originating from a sound field S surrounding the hearing device). The input unit further comprises two wireless receivers (WLR1, WLR2) for providing respective directly received auxiliary audio and / or control input signals (and / or enabling the transmission of audio and / or control signals to other devices). The hearing device HD comprises a substrate SUB on which a plurality of electronic components are mounted, including a memory MEM which for example stores different hearing aid programs (e.g. parameter settings defining the aforementioned programs, or parameters of algorithms, e.g. parameters for optimization of a neural network) and / or hearing aid configurations such as input source combinations (MEM BTE1 ,M BTE2 , WLR1, WLR2), for example optimized for a plurality of different listening situations. The substrate further comprises a configurable signal processor DSP (e.g. a digital signal processor, comprising a processor (e.g. for hearing loss compensation) HLC, a feedback suppressor FBC and a beamformer BFU and other digital functions of the hearing device according to the invention). The configurable signal processing unit DSP is adapted to access the memory MEM and select and process one or more electrical input audio signals and / or one or more directly received auxiliary audio input signals based on a currently selected (activated) hearing aid program / parameter setting (e.g. automatically selected, e.g. based on one or more sensors and / or input from a user interface). The mentioned functional units (and other elements) may be divided by circuits and components (e.g. for size, power consumption, analog-digital processing, etc.) depending on the application involved, e.g. integrated in one or more integrated circuits, or as a combination of one or more integrated circuits and one or more separate electronic components (e.g. inductors, capacitors, etc.). The configurable signal processor DSP provides a processed audio signal, which is intended to be presented to a user. The substrate also includes a front-end IC (FE) for interfacing the configurable signal processor DSP with input and output transducers, etc. and typically including interfaces between analog and digital signals. The input and output transducers can be individual separate components or integrated with other electronic circuits (e.g. based on MEMS).

[0194] The hearing device HD further comprises an output unit (eg an output transducer) for providing stimuli perceivable as sound by the user based on the processed audio signal from the processor HLC or a signal derived therefrom. Figure 5 In an embodiment of a hearing device of the type described herein, the ITE part comprises an output unit in the form of a loudspeaker (receiver) for converting an electrical signal into an acoustic (air-borne) signal which (when the hearing device is mounted at the ear of the user) is directed to the eardrum to provide a sound signal (S) there. ED The ITE part also includes a guiding element such as a dome DO for guiding and positioning the ITE part in the ear canal of the user. The ITE part also includes another input transducer such as a microphone (M ITE), used to provide a representative input sound signal (S ITE ) of the electrical input audio signal. In an embodiment, the ITE part comprises two or more input transducers configured as described in the present invention (see Figure 1A-4 , 6-8).

[0195] (From input converter M BTE1 ,M BTE2 ,M ITE The electrical input signal of can be processed according to the invention in the time domain or in the (time-)frequency domain (or partly in the time domain and partly in the frequency domain, if it is considered advantageous for the application in question). In an embodiment, one degree of freedom is used to suppress external noise and another degree of freedom is used to suppress feedback, see for example Figure 7C , 7D .

[0196] Figure 5 The hearing device HD illustrated in the example is a portable device and further comprises a battery BAT, such as a rechargeable battery, e.g. based on lithium-ion battery technology, e.g. for powering the electronic components of the BTE part and possibly the ITE part. In an embodiment, the hearing device, such as a hearing aid (e.g. the processor HLC) is adapted to provide a frequency-dependent gain and / or a level-dependent compression and / or a frequency shift of one or more frequency ranges to one or more other frequency ranges (with or without frequency compression), e.g. to compensate for a hearing impairment of a user.

[0197] Figure 6 1 shows a schematic block diagram of an embodiment of a hearing device including two microphones according to the present invention. The hearing device, such as a hearing aid, comprises a first and a second input transducer (eg, located in the ear canal, such as Figure 1Aor 3), here microphones (M1, M2), which provide respective (e.g. digitized) electrical input signals IN1, IN2 representing sounds in the user's environment. The input unit is connected via an electrical forward path to an output transducer, here a loudspeaker ("receiver") SP, for converting a processed electrical signal OUT into a stimulus perceivable as sound by the user based on the electrical input signal or a processed version thereof. The forward path comprises respective analysis filter banks (FB-A1, FB-A2) for converting respective (time domain) electrical input signals ER1, ER2 (being feedback corrected versions of the respective electrical input signals IN1, IN2) (as described below) into sub-band signals X1, X2. The forward path of the hearing device HD further comprises an adaptive beamformer filter unit BFU, which receives the sub-band signals X1, X2 and estimates EST1, EST2 of the feedback path from the output transducer to the respective first and second input transducers (as described below). The adaptive beamformer filter unit BFU is configured to provide a spatially filtered signal Y based on the electrical input signal, the feedback estimate and adaptively updated beamformer weights (e.g. based on the feedback estimate according to the invention) BF .

[0198] The hearing device further comprises a feedback estimation unit FBE which provides a feedback estimate (EST1, EST2) of a current feedback path from the output transducer SP to each input transducer (M1, M2). The hearing device is configured such that at least one of the adaptively updated beamformer weights of the adaptive beamformer filter unit BFU is updated as a function of the feedback path estimate (EST1, EST2), as proposed by the present invention. The feedback estimation unit FBE comprises respective first and second adaptive filters, each filter comprising a variable filter part (FIL1, FIL2) aiming at providing a good estimate of the "external" feedback path from (the input of) the output transducer SP to (the output of) the respective input transducer (M1, M2) and a prediction error or updating or algorithm part (ALG1, ALG2). The respective prediction error algorithm (ALG1, ALG2) uses a reference signal (here the output signal OUT) together with a signal originating from the respective microphone signal to find a setting of the adaptive filter (FIL1, FIL2) which minimizes the prediction error (by Figure 6The filter update signals UP1, UP2 in the beamformer are reflected in the filter update signals UP1, UP2 in the beamformer), when the reference signal OUT is applied to the respective adaptive filter. The estimate (EST1, EST2) of the feedback path provided by the respective adaptive filter is subtracted from the respective electrical input signal IN1, IN2 of the microphone (M1, M2) in the respective summing unit "+" to provide the so-called "error signal" (or feedback corrected signal ERR1, ERR2), which is fed to the beamformer filter unit BFU (via the respective analysis filter bank FB-A1, FB-A2) and the respective algorithm part (ALG1, ALG2) of the adaptive filter.

[0199] The hearing device HD further comprises a control unit CONT for controlling the feedback estimation unit FBE (see control signals A1ctr, A2ctr) and the beamformer filter unit BFU. The control unit CONT is for example configured to control the adaptation rate of the adaptation algorithm (e.g. defined by the time points of determining (and updating) the feedback estimate, see signals UP1, UP2). Figure 6 In an embodiment of the present invention, the control unit CONT may further comprise a detector for classifying the current acoustic environment of the user, e.g. the current feedback situation, e.g. indicating the degree of correlation between the electrical input signal (or a signal derived therefrom) and the electrical output signal. The control unit CONT may for example comprise a correlation detection unit for determining the autocorrelation of a signal of the forward path or the cross-correlation between two different signals of the forward path. The control unit CONT may further comprise other detectors, e.g. a speech detector, a feedback detector, a pitch detector, an audibility detector, a feedback change detector, etc. Preferably, the hearing device (e.g. the control unit CONT or the algorithm part (ALG1, ALG2)) comprises a memory for storing a plurality of previous estimates of the feedback path, so as to be able to rely on previous estimates when the current estimate is determined (e.g. by the control unit CONT) to be less than optimal. The control unit may store or have access via the memory MEM to a plurality of beamformer filter coefficients (see signal W). The stored beamformer filter coefficients may comprise a first set of frequency-dependent complex weighting parameters w representing the first beamformer C1. 11 (k),w 12 (k) and represents the second set of frequency-dependent complex weighting parameters w of the second beamformer C2 21 (k),w 22 (k), as above combined Figure 2 and 4 The first and second sets of weighting parameters w 11 (k),w 12 (k) and w 21 (k),w 22(k) may be predetermined, for example used as an initial value. In an embodiment, the hearing device (eg the control unit CONT) is configured to adaptively update the weighting parameter w stored in the memory during operation of the hearing device. 11 (k),w 12 (k) and w 21 (k),w 22 (k) one or more.

[0200] Furthermore, the control unit CONT may comprise a mode input for selecting a particular operating mode of the hearing device. The aforementioned mode may be selected via a user interface and / or automatically determined from a plurality of detector inputs, e.g. from a classifier of the acoustic environment, e.g. comprising one or more of an autocorrelation detector, a cross-correlation detector, a feedback detector, a voice detector, a pitch detector, a feedback change detector, an audibility detector, etc. The mode input may influence or form the basis for the control outputs A1ctr, A1ctr, HAGctr of the control unit for controlling the adaptive algorithm of the feedback estimation unit and the processing of the processor HLC. One operating mode may be a communication mode, wherein the user's own voice is picked up by a dedicated own voice beamformer and transmitted to another device, such as a telephone or a hearing device worn by another person. Such self-voice pickup may be performed instead of or in parallel with the normal operation of the beamformer filter unit, wherein the first and second microphones pick up sounds from the environment (other than the user's own voice)

[0201] The hearing device HD further comprises a processor HLC for executing one or more processing algorithms (such as compression amplification), e.g. to provide frequency-dependent gain and / or level-dependent compression and / or conversion of one or more frequency ranges to one or more other frequency ranges, e.g. to compensate for a hearing impairment of a user. Figure 6 In the embodiment of the present invention, the processor HLC receives the spatially filtered (beamformed) signal Y BF And provide the processed signal Y G , which is fed to the synthesis filter bank FB-S for processing the signal Y in K (K is, for example, 16 or 64 or more) sub-bands G The signal is converted into a processed time domain signal OUT, which is fed to an output transducer (here the speaker SP) (which may include appropriate digital-to-analog conversion circuits).

[0202] exist Figure 6 In the embodiment of the present invention, the signal processing in the analysis path (feedback estimation, etc.) is performed in the time domain. However, depending on the specific application involved, it can be performed completely or partially in the frequency domain. Figure 6 In an embodiment of the present invention, the signal processing in the forward path is performed partly in the time domain (feedback correction) and partly in the frequency domain (beamforming and hearing loss compensation).

[0203] Figure 6 The hearing devices are Figure 7B An embodiment of a slightly more general hearing device embodiment is shown in .

[0204] Fig. 7A The invention shows a method comprising two microphones (M) located in the ITE part. ITE1 ,M ITE2 ) of a hearing device HD. The ITE part comprises a housing, two ITE microphones (M ITE1 ,M ITE2 ) is located therein (e.g. in the longitudinal direction of the shell along the ear canal axis (see Fig. 7A The dotted arrow "inwards")) when the hearing device HD is mounted on or at the user's ear during operation. The ITE part also includes a guide ( Fig. 7A The ITE part also includes a speaker (towards the eardrum) for playing the resulting audio signal to the user, thereby generating a sound field in the residual cavity. A portion of it is directed toward the ITE microphone (M ITE1 ,M ITE2 ) and the environment. The hearing device (eg an ITE part, which may constitute a part customized for the user's ear, for example in terms of shape, or alternatively have a standardized shape) comprises a plurality of different functional modules (BFU, HLC, FBE). Figure 7B Shown as Fig. 7A A schematic block diagram of an embodiment of a hearing device is shown in FIG. The loudspeaker SP, the beamformer filter unit BFU, the processor HLC and the feedback estimation unit FBE have a combined Figure 6 The hearing device HD may be configured to be located in the soft part of the ear canal of the user. In an embodiment, the hearing device HD is configured to be located completely or partially in the bony part of the ear canal.

[0205] Figure 7C An embodiment of a hearing device according to the invention comprising three microphones in the ITE part is shown. Fig.7D Shown as Figure 7C A schematic block diagram of an embodiment of a hearing device is shown in . Figure 7C and 7D An embodiment of a hearing device HD comprises three microphones (M ITE11 ,M ITE12 ,M ITE2 ). Two of these microphones (M ITE11 ,M ITE12 ) towards the environment, and a microphone (M ITE2) towards the eardrum (when the hearing device is mounted in place in operation). The hearing device comprises or consists of an ITE part comprising a seal for providing a tight seal against the ear canal wall (see Figure 7C The "seal" in the figure is used to acoustically "isolate" the microphone M facing the eardrum. ITE2 Protection from ambient sound incident on the ear canal (and hearing device) ITE The impact of Figure 7C The hearing device HD comprises Fig. 8A and 8B The same functional elements of the embodiment. Fig.7D The embodiment further comprises a corresponding feedback cancellation system (including a combining unit "+" for subtracting the beamforming signal Y BF The microphone signals IN11, IN12 towards the environment are fed to a first beamformer unit BFU1 to provide a first (far field) beamformed signal Y BF1 The "directional microphone" (microphone facing forward) ITE11 ,M ITE12 ) and the first beamformer unit BFU1) is represented by the estimate ESTBF of the feedback path from the first (far-field) beamforming signal Y BF1 The subtraction provides a feedback-corrected beamforming signal ERBF which is fed to the second beamformer unit BFU2. ITE2 The signal IN2 is connected to the combined unit "+", from the loudspeaker SP to the microphone M towards the eardrum. ITE2 The estimate of the feedback path of is subtracted, which provides a feedback-corrected microphone signal ER2 towards the eardrum. This signal is fed to the second beamformer unit BFU2, which provides a resulting far-field and feedback-minimized beamformed signal Y based on the input signal (ERBF, ER2) and the feedback estimate (ESTBF, EST2). BF The resulting beamformed signal Y BF In the processor HLC, the signal is subjected (or may be subjected) to one or more processing algorithms (e.g. compression amplification to compensate for the hearing loss of the user). The resulting processed signal OUT is fed to the output transducer (speaker SP) and played to the user as a sound signal. The resulting processed signal OUT is also fed to the feedback estimation unit FBE as a reference signal.

[0206] Fig. 7E An embodiment of a hearing device HD according to the invention is shown comprising four microphones, two of which (M BTE1 ,M BTE2 ) is located in the BTE section (BTE) and two microphones (MITE1 ,M ITE2 ) is located in the ITE part (ITE). The BTE part is adapted to be located at or behind the ear (pinna), and the ITE part is adapted to be located at or in the ear canal (of the same ear) of the user. The BTE part and the ITE part are electrically connected (via wires or wirelessly). The ITE part comprises a housing, two ITE microphones (M ITE1 ,M ITE2 ) is located therein (e.g. in the longitudinal direction of the shell along the ear canal axis (see Fig. 7E The dotted arrow "inwards")) when the hearing device HD is mounted on or at the user's ear during operation. The ITE part also includes a guide ( Fig. 7E The ITE part further comprises a loudspeaker (towards the eardrum) for playing the resulting audio signal to the user, thereby generating a sound field SED in the residual cavity. A portion of it is directed toward the ITE microphone (M ITE1 ,M ITE2 ) and the environment. The BTE part includes a housing, two BTE microphones (M BTE1 ,M BTE2 ) are located therein (e.g. at the top of the housing so that they lie in a horizontal plane when properly mounted at the user's ear) (so that the microphone axis is parallel to the user's viewing direction, see Fig. 7E ).

[0207] Figure 7F Shown as Fig. 7E Schematic block diagram of an embodiment of a hearing device shown in . The hearing device (such as a BTE part and / or an ITE part) comprises a processing unit configured to process microphone signals according to the present invention (see Figure 7F The hearing device further comprises a unit FBE, BFU, HLC in the hearing device, comprising estimating and minimizing feedback from the loudspeaker SP to the microphone and applying suitable beamforming to the microphone signal (at least in a certain operating mode). The hearing device further comprises a processor HLC for applying a relevant processing algorithm to the (possibly) beamformed signal Y BF The processed signal OUT from the processor HLC is fed to the loudspeaker SP for presentation to the user and to the feedback estimation unit FBE as a reference signal.

[0208] like Figure 7F As shown in the figure, the ITE microphone (M ITE1 ,M ITE2 ) receives the sound field S including feedback from nearby speakers ITE And provide ITE microphone signal (IN ITE1 ,IN ITE2), which is fed to the corresponding combination unit ('+'), where the corresponding feedback estimate (EST ITE1 ,EST ITE2 ) is subtracted to provide the feedback-corrected ITE microphone signal (ER ITE1 ,ER ITE2 The (feedback corrected) microphone signal from the ITE microphone is used in the beamformer filter unit BFU to provide one or more beamformers for cancelling or minimizing the resulting beamformed signal Y BF Feedback in .

[0209] like Figure 7F As shown in Figure 2, the BTE microphone (M BTE1 ,M BTE2 ) receives a sound field S including less feedback than an ITE microphone BTE And provide BTE microphone signal (IN BTE1 ,IN BTE2 ), which is fed to the corresponding combination unit ('+'), where the corresponding feedback estimate (EST BTE1 ,EST BTE2 ) is subtracted to provide the feedback-corrected BTE microphone signal (ER BTE1 ,ER BTE2 ). The (feedback-corrected) BTE microphone signal (IN BTE1 ,IN BTE2 ) is used in a beamformer filter unit BFU to provide one or more beamformers directed towards the environment (such as a nearby speaker or a user's mouth).

[0210] The feedback estimation unit FBE is configured to provide feedback from the loudspeaker SP to the four microphones (M BTE1 ,M BTE2 ,M ITE1 ,M ITE2 ) is the corresponding estimate (EST) of the feedback path of each BTE1 ,EST BTE2 ,EST ITE1 ,EST ITE2 The feedback estimator is based on the corresponding feedback-corrected input signal (ER BTE1 ,ER BTE2 ,ER ITE1 ,ER ITE2 ), the processed output signal OUT and the weight WGT applied in the possible beamformer filter unit BFU, for example, see the combination Figures 8A-8B Discussions taking place.

[0211] Generally, microphones located in the BTE part are good at extracting ambient noise from the background, while microphones located in the ITE part are good at extracting feedback. Figure 5 or 7E, 7F hearing devices may be configured to use a BTE microphone (e.g. Fig. 7E , 7F M BTE1 ,M BTE2 ) is used to estimate the post-filter gain to obtain the BTE microphone signal (e.g. Figure 7F IN BTE1 ,IN BTE2 ) reduces noise in a beamformer such as a target cancellation beamformer. The post-filter gain may be applied, for example, to a signal in a forward path of a hearing device, wherein the signal in the forward path is based on a feedback cancellation beamformer based on two BTE microphone signals (such as Figure 7F IN BTE1 ,IN BTE2 ) or based on ITE microphone signals (such as Figure 7F IN ITE1 ,IN ITE2 ) or a combination of a BTE microphone signal and an ITE microphone signal. Such a configuration combines Fig.9A , 9B , 9C for further discussion.

[0212] Fig. 7A , 7C The embodiments of 7E and 7E may represent processing in the time domain, but may alternatively include corresponding filter banks to provide processing in the (time-)frequency domain (e.g. based on a short-time Fourier transform (STFT)), see e.g. Figure 6 and Fig.9A , 9B , 9C includes an embodiment of the corresponding analysis and synthesis filter group.

[0213] example

[0214] In the previous example, two microphones oriented along the axis from the opening of the outer ear towards the ear drum into the ear canal have been included. The signal from this microphone pair is subjected to a beamformer, which is adjusted to process far-field sounds originating from outside the ear just like in a single omnidirectional microphone system, while suppressing feedback signals received by the directional microphone system (which are generated in the near field). Thus, ultra-high feedback suppression is possible while receiving far-field sounds from the surroundings in much the same way as in a single-microphone hearing instrument.

[0215] Therefore, the present invention utilizes additional anti-feedback properties, which can be combined from the above Figure 1A-Figure 4 , Figure 6The spatial signal separation described for a two-microphone system is achieved. In the following further embodiments, these principles are applied in a system with three microphones, two of which represent a conventional directional system as described above, and where the third microphone is added for spatial feedback suppression purposes.

[0216] Fig. 8A An embodiment of a hearing device according to the invention comprising three microphones in the ITE part is shown.

[0217] Figure 8B Shown as Fig. 8A A schematic block diagram of an embodiment of a hearing device is shown in .

[0218] The proposed hearing instrument is configured in Fig. 8A The hearing instrument HD comprises an ITE part (ITE) comprising three input transducers, here microphones. An "external microphone" (M) positioned towards the environment, for example at the ear canal opening (e.g. in the housing of the ITE part) ITE11 ,M ITE12 ) provides directional information to enhance speech intelligibility of the target signal (and may contribute to the reduction of noise from the environment). ITE2 , positioned closest to the eardrum (see the hatched ellipse labeled "eardrum" and the dotted arrow labeled "inward" indicating the direction toward the inner ear / eardrum)) serves as a means of obtaining spatial anti-feedback information to increase the audiological performance in terms of acoustic amplification. Preferably, the ITE portion includes a seal toward the ear canal wall so that the ITE portion fits tightly with the ear canal wall (or at least provides a controlled or minimal sound leakage path). The ITE portion may include a vent to minimize the occlusion effect. The purpose of the seal may further minimize the access to the inner microphone (M ITE2 ) minimizes the ambient noise in the sound field, when the internal microphone (M ITE2 ) signal and the external microphone (M ITE11 ,M ITE12 , see for example Figure 8B ) signals to avoid (re)introducing ambient noise into the beamformed signal.

[0219] The spatial anti-feedback feature can be implemented as a spatial feedback system, see the beamformer filter unit ( Figure 8B The dotted box in the figure is marked as BFU), which consists of an internal microphone (M ITE2 ) and used as a microphone (see Figure 8B Signal Y in FF ) external microphone pair (M ITE11 ,M ITE12). In this embodiment, the output signals from the two outer microphones can be averaged as a means of obtaining spatial anti-feedback for both microphones using only one anti-feedback system. As an alternative, the performance is further enhanced by using two separately optimized spatial anti-feedback systems. In this embodiment, two sets of optimizations are performed, one for microphone M and one for microphone M. ITE11 and M ITE2 (See Fig. 8A ) and a set of microphones M ITE12 and M ITE2 .

[0220] If we take the external microphone (M ITE11 ,M ITE12 ) as a single microphone element, we assume that the microphone system has a joint feedback path. However, if we have an adaptive microphone system, the resulting joint feedback path will change according to the directional weights. If we know the estimates of the two outer acoustic feedback paths (h1,h2 (impulse response) or H1,H2 (frequency response)) and the directional weights (w1,w2), we can calculate the joint outer feedback path, which we can then use together with the feedback path of the inner microphone (as described below) to adjust the directivity pattern.

[0221] In the case of a beamformer filter unit BFU representing an adaptive directional system, two external ITE microphones (M ITE11 ,M ITE12 ) will vary according to the adaptive directional system. h1 and h2 are the impulse responses of the acoustic feedback path, and w1 and w2 are the adaptive weights of the directional system (BFU1, which can also be implemented in the frequency domain).

[0222] Since the joint adaptive system is given by w1*h1+w2*h2, the (joint) feedback path can be varied only according to the adaptation parameters of the directional system (ie h1 and h2 are kept constant).

[0223] The adaptive weights (or impulse responses) (w3 and w4) of the directional feedback cancellation system will therefore be adjusted according to this change and may therefore depend on w1, w2 and the (fixed or adaptive) estimates of the feedback path (h1, h2 and h3).

[0224] Fig.9A , 9B 9C show three different embodiments of hearing devices according to the present invention. According to one aspect of the present invention, each hearing device HD comprises two input transducers (here microphones M1, M2) for canceling noise in the environment and feedback from an output transducer (here loudspeaker SP, for example) to the input transducers (M1, M2). Fig.9A , 9BEach of the embodiments of 9C includes a microphone array comprising at least two microphones (M1, M2) positioned such that the microphone array can be used to cancel external noise and feedback. The at least two microphones may, for example, include two BTE microphones (e.g., arranged Fig. 7E M BTE1 ,M BTE2 ) or two ITE microphones (e.g. set up as Figure 7C M ITE11 ,M ITE12 ) or two BTE microphones (e.g. set up as Fig. 7E M BTE1 ,M BTE2 ) and an ITE microphone (e.g. set up as Figure 5 M ITE or Figure 7C M ITE2 ) or three ITE microphones (e.g. Figure 7C ).

[0225] Fig.9A A first embodiment of a hearing device HD is shown, comprising two microphones (M1, M2) for cancelling noise in the environment and feedback from a loudspeaker SP to the microphones (M1, M2). The microphone signals (x1, x2) are propagated through respective analysis filter banks FBA to obtain frequency domain representations (X1, X2) of the two microphone signals. The frequency domain microphone signals are processed in two beamformer units (BFU1 and BFU2). The first beamformer unit has two output signals: C1 for (possibly adaptive) enhancement of target sounds from a given direction and a target cancellation beamformer C2 for cancelling sounds from a given target direction. The two directional signals are propagated into a post-filter module PF for estimating the signal-to-noise ratio, which is converted into a gain G, which varies across time and frequency (G=G(k,m), where k and m are frequency and time indices, respectively, see e.g. EP2701145A1). The gain is multiplied to the output Y of the other beamformer unit BFU2 BF2 , which generates a (possibly adaptive) directional signal Y that aims to cancel out feedback and ambient noise BF The resulting signal is converted back to a time domain signal OUT using a synthesis filter bank AFS and presented to the listener. Thereby, while the directional signal aims at eliminating feedback, the post-filter gain aims at eliminating external noise.

[0226] Fig. 9B A second embodiment of a hearing device HD is shown comprising two input transducers (M1, M2) for cancelling noise in the environment and feedback from the output transducer (SP) to the input transducers (M1, M2). Fig. 9B Examples and Fig.9AThe embodiment of is similar to that of , except that it comprises only one beamformer unit BFU receiving an electrical (sub-band) input signal (X1, X2) from a microphone. The beamformer unit BFU provides a beamformer C1 which (possibly adaptively) enhances a target sound coming from a given direction. The post-filter PF converts xx into a gain G while attenuating the "noise" from the feedback path. The resulting gain G is applied to the target signal C1 (see multiplication unit "x"), thereby providing a resulting beamformed signal, which is converted into a time domain signal OUT in a synthesis filter bank SFB and fed to a loudspeaker SP for presentation to the eardrum of the user. The directional signal C1 aims to cancel the noise in the external sound, and the post-filter gain G aims to cancel the feedback signal. In this case, the noise estimate can be the feedback signal (see the input signals FB1, FB2 of the post-filter FP) (or a single feedback estimate, or a combination (such as a MAX value), instead of a target cancellation beamformer (such as Fig.9A C2)).

[0227] Fig. 9C A third embodiment of a hearing device HD is shown comprising two input transducers (M1, M2) for cancelling noise in the environment and feedback from the output transducer (SP) to the input transducers (M1, M2). Fig. 9C The embodiment is equivalent to Fig. 9B but Fig. 9C The beamformer unit BFU in is updated by the corresponding feedback path estimates (FB1, FB2) from the loudspeaker SP to the microphones (M1, M2). Fig. 9C In the embodiment of , the directional system BFU and the post filter PF are adjusted to minimize feedback (see input signals (FB1, FB2)).

[0228] exist Fig.9A , 9B , 9C hearing device embodiment, the spatially filtered (beamformed) and noise-reduced signal Y BF Of course, it may be subjected to other processing algorithms (such as compression amplification to compensate for the user's hearing loss) before being presented to the user (see, for example Figure 6 or Figure 7B , 7D , processor HLC in 7F).

[0229] The structural features of the apparatus described above, described in detail in the “Detailed Description of the Invention” and defined in the claims may be combined with the steps of the method of the present invention when appropriately replaced by corresponding processes.

[0230] Unless expressly stated, the singular forms "one", "the" used herein include the plural form (i.e., have the meaning of "at least one"). It should be further understood that the terms "having", "including" and / or "comprising" used in the specification indicate the presence of the described features, integers, steps, operations, elements and / or parts, but do not exclude the presence or addition of one or more other features, integers, steps, operations, elements, parts and / or combinations thereof. It should be understood that, unless expressly stated, when an element is referred to as being "connected" or "coupled" to another element, it may be directly connected or coupled to the other element, or there may be one or more intermediate intervening elements. As used herein, the term "and / or" includes any and all combinations of one or more listed related items. Unless expressly stated, the steps of any method disclosed herein do not have to be performed in the exact order disclosed.

[0231] It should be appreciated that reference to "an embodiment" or "embodiment" or "aspect" or features that "may" be included in this specification means that the specific features, structures or characteristics described in conjunction with the embodiment are included in at least one embodiment of the present invention. In addition, the specific features, structures or characteristics may be appropriately combined in one or more embodiments of the present invention. The foregoing description is provided to enable those skilled in the art to implement the various aspects described herein. Various modifications will be apparent to those skilled in the art, and the general principles defined herein may be applied to other aspects.

[0232] The claims are not limited to the various aspects shown herein, but rather have the full scope consistent with the claim language wherein, unless expressly stated otherwise, elements referred to in the singular do not mean "one and only one" but rather "one or more." Unless expressly stated otherwise, the term "some" means one or more.

[0233] Therefore, the scope of the present invention should be judged based on the following claims.

[0234] References

[0235] ·EP2843971A1(OTICON)04.03.2015

[0236] ·EP2701145A1(RETUNE DSP,OTICON)26.02.2014

[0237] ·EP3253075A1(OTICON)06.12.2017

Claims

1. A hearing device configured to be located at or in an ear of a user, the hearing device comprising an ITE part adapted to be located at or in an ear canal of a user, the ITE part comprising: - a housing configured to be located at least partially in an ear canal of a user; at least two outer input transducers and at least one inner input transducer for providing respective electrical input signals, wherein when the ITE part is mounted at or in the ear canal in operation, the at least two outer input transducers face the environment and provide respective electrical input signals representative of sounds in the user's environment, and the at least one inner input transducer is closer to the eardrum than the at least two outer input transducers, the at least one inner input transducer providing at least one electrical input signal representative of sounds reflected from the eardrum; - a beamformer filter unit connected to the at least two outer input transducers and the at least one inner input transducer, wherein the beamformer filter unit comprises a first beamformer for spatially filtering sounds in the environment based on respective electrical input signals from the at least two outer input transducers towards the environment and respective beamformer weights and providing a first spatially filtered signal; and a second beamformer for spatially filtering the sound reflected from the eardrum based on at least one electrical input signal from the at least one medial input transducer towards the eardrum, the first spatially filtered signal, and corresponding beamformer weights and providing a second spatially filtered signal; - an output transducer for providing a stimulus perceptible as sound by an eardrum of a user based on said second spatially filtered signal.

2. The hearing device according to claim 1, configured such that the first and second beamformers are simultaneously available.

3. The hearing device according to claim 1, wherein: The first and / or second beamformer is adaptive.

4. The hearing device according to claim 1, wherein: The housing comprises a seal towards the ear canal wall so that the ITE part fits closely to the ear canal wall or at least provides a controlled or minimal sound leakage path.

5. The hearing device according to claim 4, wherein: The at least two outboard input transducers and the at least one inboard input transducer are located on each side of the seal.

6. The hearing device according to claim 1 , configured to provide each of the respective electric input signals as a subband signal X in a time-frequency representation (k, m) i (k,m), i=1,…,M, where M is the number of input transformers, k and m are frequency and time indices, respectively, and k=1,…,K, where K is the number of subbands.

7. The hearing device according to claim 1, wherein: The beamformer weights for different channels serve different purposes.

8. The hearing device according to claim 7, configured such that in channels where feedback is dominant, the directional system is used for feedback cancellation, and in channels where feedback is not significant, the directional system is used for noise reduction of external noise sources or microphone noise.

9. The hearing device according to claim 1, wherein: The ITE portion includes vents to minimize occlusion effects.

10. The hearing device according to claim 4, wherein: At least two outer microphones are located outside the seal and facing the environment, and at least one inner microphone is located inside the seal and facing the ear drum.

11. The hearing device according to claim 1, comprising a feedback estimation unit providing a feedback estimate of a current feedback path from the output transducer to each of the at least two outer input transducers and at least one inner input transducer.

12. The hearing device according to claim 1, comprising a signal processor for enhancing an input signal and providing a processed output signal.

13. The hearing device according to claim 1, adapted to provide frequency dependent gain and / or level dependent compression and / or frequency transposition of one or more frequency ranges to one or more other frequency ranges to compensate for hearing impairment of a user.

14. The hearing device according to claim 1, configured to pick up the user's own voice via a predetermined or adaptive beamformer focused on the user's mouth.

15. The hearing device of claim 1, comprising a self-speech beamformer focused on a user's mouth and an ambient sound beamformer focused on sound sources of interest in the user's environment, the self-speech beamformer and the ambient sound beamformer being created simultaneously using the electrical input signal.

16. The hearing device according to claim 1, wherein: The at least two outer input transducers include first and second outer microphones facing toward the environment, the at least one inner input transducer includes a third microphone facing toward the eardrum, Wherein, the first beamformer comprises - two adaptively determined weighting units for applying respective weights to the two electric input signals from the outer microphones and providing first and second weighted electric input signals; and - a first summing unit for adding the first and second weighted electrical input signals to provide a first spatially filtered signal; Wherein, the second beamformer comprises - two weighting units for adaptively determining and applying respective weights to the first spatially filtered signal and the electrical signal from the inner microphone to provide third and fourth weighted electrical input signals; and - a second summing unit for adding the third and fourth weighted electrical input signals to provide a second spatially filtered signal; The output transducer is a speaker, the output of which is connected to the output of the second summing unit.

17. The hearing device according to claim 1, wherein: The hearing device is a hearing aid, a headset, an ear protection device or a combination thereof.

Citation Information

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