Hearing aid comprising a feedback control system

Abstract

A hearing aid includes a feedback control system for handling external feedback from an output transducer to an input transducer. The feedback control system includes an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter configured to provide a current estimate of the feedback path transfer function; a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of the forward path transfer function, the instant open loop gain estimate; and an adaptive filter controller for providing an update transfer function estimate for the adaptive filter in dependence of the instant estimate of the feedback path transfer function. The hearing aid is configured to use the update transfer function estimate in the adaptive filter to update the current estimate of the feedback path transfer function. A method of detecting a sudden change in a feedback/echo path is further disclosed.

Claims

1. A hearing aid configured to be worn by a user, the hearing aid comprising a forward path comprising: an input transducer configured to convert sound in an environment of the user to an electric input signal representing said sound, a processor for processing said electric input signal (X) or a signal derived therefrom and for providing a processed signal; an output transducer for converting said processed signal to stimuli perceivable by the user as sound; the forward path providing a forward path transfer function F(k,n), where k and n are frequency and time indices, respectively, a feedback control system for handling external feedback from the output transducer to the input transducer, the feedback control system comprising: an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter comprising an adaptive algorithm configured to provide a current estimate of a feedback path transfer function, and a variable filter configured to provide an estimate of the current feedback signal from the output transducer to the input transducer based on said current estimate of the feedback path transfer function and the processed signal; a combination unit configured to subtract said current estimate of the feedback signal from said electric input signal, or a processed version thereof, and to provide a feedback corrected signal, termed the error signal, a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of said forward path transfer function F(k,n), said instant open loop gain estimate, and optionally said current estimate of the feedback path transfer function; and an adaptive filter controller for providing an update transfer function estimate for said adaptive filter in dependence of said instant estimate of the feedback path transfer function, wherein the hearing aid is configured to provide that the update transfer function estimate is used in the adaptive filter to update the current estimate of the feedback path transfer function, and said update transfer function estimate is equal to said instant estimate of the feedback path transfer function.

2. A hearing aid according to claim 1 wherein the adaptive algorithm comprises an LMS, or an NLMS algorithm.

3. A hearing aid according to claim 1 wherein said adaptive algorithm comprises an NLMS algorithm, and wherein a residual feedback path transfer function is estimated by the NLMS algorithm, the estimate of the residual feedback path transfer function is defined as the difference between the estimate of the feedback path transfer function after a sudden change of the feedback path and the estimate of the feedback path transfer function before the sudden change occurred, the latter being given by the current estimate of the feedback path transfer function provided by the adaptive algorithm.

4. A hearing aid according to claim 1 comprising one or more analysis filter banks allowing one or more signals of the hearing aid to be processed in a time-frequency domain.

5. A hearing aid according to claim 1 comprising a feedback instability detector for monitoring the fulfillment of a feedback path instability criterion.

6. A hearing aid according to claim 1 being constituted by or comprising an air-conduction type hearing aid, a bone-conduction type hearing aid, or a combination thereof.

7. A hearing aid configured to be worn by a user, the hearing aid comprising a forward path comprising: an input transducer configured to convert sound in an environment of the user to an electric input signal representing said sound, a processor for processing said electric input signal (X) or a signal derived therefrom and for providing a processed signal; an output transducer for converting said processed signal to stimuli perceivable by the user as sound; the forward path providing a forward path transfer function F(k,n), where k and n are frequency and time indices, respectively, a feedback control system for handling external feedback from the output transducer to the input transducer, the feedback control system comprising: an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter comprising an adaptive algorithm configured to provide a current estimate of a feedback path transfer function, and a variable filter configured to provide an estimate of the current feedback signal from the output transducer to the input transducer based on said current estimate of the feedback path transfer function and the processed signal; a combination unit configured to subtract said current estimate of the feedback signal from said electric input signal, or a processed version thereof, and to provide a feedback corrected signal, termed the error signal, a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of said forward path transfer function F(k,n), said instant open loop gain estimate, and optionally said current estimate of the feedback path transfer function; and an adaptive filter controller for providing an update transfer function estimate for said adaptive filter in dependence of said instant estimate of the feedback path transfer function, wherein the hearing aid is configured to provide that the update transfer function estimate is used in the adaptive filter to update the current estimate of the feedback path transfer function, and said feedback change estimator is configured to provide said update transfer function estimate as a linear combination of said instant open loop gain estimate divided by said forward path transfer function (F(k,n)) and said current estimate of the feedback path transfer function.

8. A hearing aid configured to be worn by a user, the hearing aid comprising a forward path comprising: an input transducer configured to convert sound in an environment of the user to an electric input signal representing said sound, a processor for processing said electric input signal (X) or a signal derived therefrom and for providing a processed signal; an output transducer for converting said processed signal to stimuli perceivable by the user as sound; the forward path providing a forward path transfer function F(k,n), where k and n are frequency and time indices, respectively, a feedback control system for handling external feedback from the output transducer to the input transducer, the feedback control system comprising: an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter comprising an adaptive algorithm configured to provide a current estimate of a feedback path transfer function, and a variable filter configured to provide an estimate of the current feedback signal from the output transducer to the input transducer based on said current estimate of the feedback path transfer function and the processed signal; a combination unit configured to subtract said current estimate of the feedback signal from said electric input signal, or a processed version thereof, and to provide a feedback corrected signal, termed the error signal, a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of said forward path transfer function F(k,n), said instant open loop gain estimate, and optionally said current estimate of the feedback path transfer function; and an adaptive filter controller for providing an update transfer function estimate for said adaptive filter in dependence of said instant estimate of the feedback path transfer function, wherein the hearing aid is configured to provide that the update transfer function estimate is used in the adaptive filter to update the current estimate of the feedback path transfer function, and said open loop gain estimator is configured to provide said instant open loop gain estimate as
{circumflex over (L)}.sub.fast(k, n)=E(k, n)/E(k,n−D) where E(k,n) is the error signal at time instance n and E(k,n-D) is the error signal one loop delay D, or an estimate thereof, earlier, and where the loop delay D represents a roundtrip delay of the audio path of the hearing aid.

9. A hearing aid configured to be worn by a user, the hearing aid comprising a forward path comprising: an input transducer configured to convert sound in an environment of the user to an electric input signal representing said sound, a processor for processing said electric input signal (X) or a signal derived therefrom and for providing a processed signal; an output transducer for converting said processed signal to stimuli perceivable by the user as sound; the forward path providing a forward path transfer function F(k,n), where k and n are frequency and time indices, respectively, a feedback control system for handling external feedback from the output transducer to the input transducer, the feedback control system comprising: an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter comprising an adaptive algorithm configured to provide a current estimate of a feedback path transfer function, and a variable filter configured to provide an estimate of the current feedback signal from the output transducer to the input transducer based on said current estimate of the feedback path transfer function and the processed signal; a combination unit configured to subtract said current estimate of the feedback signal from said electric input signal, or a processed version thereof, and to provide a feedback corrected signal, termed the error signal, a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of said forward path transfer function F(k,n), said instant open loop gain estimate, and optionally said current estimate of the feedback path transfer function; and an adaptive filter controller for providing an update transfer function estimate for said adaptive filter in dependence of said instant estimate of the feedback path transfer function, wherein the hearing aid is configured to provide that the update transfer function estimate is used in the adaptive filter to update the current estimate of the feedback path transfer function, the hearing aid comprises a feedback instability detector for monitoring the fulfillment of a feedback path instability criterion, and said feedback path instability detector is configured to determine current gradient values of the adaptive algorithm to adapt one or more of the current filter coefficients of the adaptive filter and to provide smoothed and possibly further processed, versions thereof, and wherein said instability criterion comprises a comparison of said current gradient values to one or more threshold values.

10. A method of operating a hearing aid, the hearing aid being configured to be worn by a user, the hearing aid comprising a forward path comprising: an input transducer configured to convert sound in an environment of the user to an electric input signal representing said sound, a processor for processing said electric input signal or a signal derived therefrom and for providing a processed signal; an output transducer for converting said processed signal to stimuli perceivable by the user as sound; wherein the method comprises providing a forward path transfer function F(k,n), where k and n are frequency and time indices, respectively, handling external feedback from the output transducer to the input transducer by providing an instant open loop gain estimate; adaptively providing, by an adaptive algorithm, a current estimate of a feedback path transfer function, adaptively providing, by an adaptive filter with filter coefficients determined by said adaptive algorithm, an estimate of the current feedback signal from the output transducer to the input transducer based on said current estimate of the feedback path transfer function and the processed signal; subtracting said current estimate of the feedback signal from said electric input signal, or a processed version thereof, to provide a feedback corrected signal, termed the error signal, providing an instant estimate of the feedback path transfer function in dependence of said forward path transfer function F(k,n), said instant open loop gain estimate and optionally of said current estimate of the feedback path transfer function; adaptively providing an update transfer function estimate in dependence of said instant estimate of the feedback path transfer function; providing that the update transfer function estimate is used in the adaptive filter to update the current estimate of the feedback path transfer function, wherein said update transfer function estimate is equal to said instant estimate of the feedback path transfer function.

11. A method according to claim 10 comprising, monitoring the fulfillment of a feedback path instability criterion.

12. A method according to claim 11 wherein the instability criterion is based on magnitude, phase, or derivatives of magnitude and phase of the electric input signal, or a signal derived therefrom.

13. A method according to claim 11 wherein the instability criterion is based on comparing smoothed versions of one or more gradient values gradient of the adaptive algorithm to a threshold value.

14. A method according to claim 13 wherein the instability criterion is fulfilled when the one or more gradient values or a weighted combination of said one or more gradient values are or is larger than the threshold value.

15. A method according to claim 10 comprising determining current gradient values of the adaptive algorithm to adapt one or more of the current filter coefficients of the adaptive filter and to provide smoothed and possibly further processed, versions thereof.

16. A non-transitory computer-readable medium storing a computer program comprising instructions which, when the program is executed by a computer, cause the computer to carry out the method of claim 10.

Description

BRIEF DESCRIPTION OF DRAWINGS

(1) The aspects of the disclosure may be best understood from the following detailed description taken in conjunction with the accompanying figures. The figures are schematic and simplified for clarity, and they just show details to improve the understanding of the claims, while other details are left out. Throughout, the same reference numerals are used for identical or corresponding parts. The individual features of each aspect may each be combined with any or all features of the other aspects. These and other aspects, features and/or technical effect will be apparent from and elucidated with reference to the illustrations described hereinafter in which:

(2) FIG. 1A shows a first embodiment of a hearing aid comprising a feedback control system according to the present disclosure, and

(3) FIG. 1B shows a second embodiment of a hearing aid comprising a feedback controlsystem according to the present disclosure,

(4) FIG. 2 shows a flowchart describing a scheme for updating a feedback estimate of a feedback control system according to the present disclosure,

(5) FIG. 3 shows the feedback loop of a hearing aid comprising an electric forward path from input to output transducer, and an acoustic (and/or mechanical) feedback loop from output to input transducer,

(6) FIG. 4 schematically shows an exemplary time dependence of a true feedback path and an estimated feedback path according to the present disclosure,

(7) FIG. 5 shows a flow diagram for a method of detecting a sudden change of a feedback/echo path of a hearing aid or headset, and

(8) FIG. 6 shows exemplary waveforms of signals from which the sudden change of feedback/echo path can be identified according to a method of the present disclosure,

(9) FIG. 7 schematically illustrates a hearing aid according to the present disclosure when located in an ear canal close to the eardrum of a user, and

(10) FIG. 8 shows a schematic drawing of an exemplary feedback cancellation system of a hearing aid.

(11) The figures are schematic and simplified for clarity, and they just show details which are essential to the understanding of the disclosure, while other details are left out. Throughout, the same reference signs are used for identical or corresponding parts.

(12) Further scope of applicability of the present disclosure will become apparent from the detailed description given hereinafter. However, it should be understood that the detailed description and specific examples, while indicating preferred embodiments of the disclosure, are given by way of illustration only. Other embodiments may become apparent to those skilled in the art from the following detailed description.

DETAILED DESCRIPTION OF EMBODIMENTS

(13) The detailed description set forth below in connection with the appended drawings is intended as a description of various configurations. The detailed description includes specific details for the purpose of providing a thorough understanding of various concepts. However, it will be apparent to those skilled in the art that these concepts may be practiced without these specific details. Several aspects of the apparatus and methods are described by various blocks, functional units, modules, components, circuits, steps, processes, algorithms, etc. (collectively referred to as “elements”). Depending upon particular application, design constraints or other reasons, these elements may be implemented using electronic hardware, computer program, or any combination thereof.

(14) The electronic hardware may include micro-electronic-mechanical systems (MEMS), integrated circuits (e.g. application specific), microprocessors, microcontrollers, digital signal processors (DSPs), field programmable gate arrays (FPGAs), programmable logic devices (PLDs), gated logic, discrete hardware circuits, printed circuit boards (PCB) (e.g. flexible PCBs), and other suitable hardware configured to perform the various functionality described throughout this disclosure, e.g. sensors, e.g. for sensing and/or registering physical properties of the environment, the device, the user, etc. Computer program shall be construed broadly to mean instructions, instruction sets, code, code segments, program code, programs, subprograms, software modules, applications, software applications, software packages, routines, subroutines, objects, executables, threads of execution, procedures, functions, etc., whether referred to as software, firmware, middleware, microcode, hardware description language, or otherwise.

(15) The present application relates to the field of hearing aids, in particular to feedback control. Feedback estimation may be provided by an adaptive filter comprising a variable filter whose transfer function (e.g. governed by filter coefficients) can be dynamically updated to estimate a feedback path from an output transducer to an input transducer. The dynamic determination and update of the transfer function may be generally handled by an adaptive algorithm, such as an LMS or NLMS algorithm as is known in the art. By sudden changes of the feedback path, however, there may be a need for a more instant (event driven) determination and update of the transfer function (e.g. to enhance convergence of the adaptive algorithm).

(16) We propose a general method of improving the convergence/tracking abilities of the adaptive filter, by using an estimation of the true open loop transfer function L(k,n), at the frequency index k and time index n, given by
L(k,n)=H.sub.res(k,n).Math.F(k,n),  (1)
where H.sub.res(k,n)=(H(k,n)−Ĥ(k,n)), and where H(k,n) denotes the transfer function for the unknown feedback path, Ĥ(k,n) denotes the transfer function of the estimated feedback path, and F(k,n) is the known forward path transfer function in a hearing aid.

(17) In a traditional feedback cancellation system, the adaptive filter provides a feedback path estimate (based on an estimate Ĥ(k,n) of the feedback path transfer function). However, it has limited convergence/tracking abilities in dynamic feedback situations.

(18) Embodiments of a hearing aid comprising a feedback control system (FBC, as e.g. illustrated in FIG. 1A) according to the present disclosure are illustrated in FIGS. 1A and B.

(19) The embodiments of a hearing aid (HA) of FIGS. 1A and 1B both comprise a forward path for processing an audio sound signal (‘Acoustic input’). The audio sound signal may comprise a mixture of sound s.sub.x of origin external to the hearing aid (e.g. speech ad noise) and feedback sound v from an output transducer (OT) to an input transducer (IT) of the hearing aid. The feedback path (FBP) from the output transducer to the input transducer has a (frequency) transfer function H. The forward path comprises the input transducer (IT) configured to convert sound in an environment of the user to an electric input signal (X) representing the audio sound signal (where X=S.sub.x+V, S.sub.x and V being the electric (possibly digitized, possibly.sup.- :frequency domain) equivalents of sound signals s.sub.x and v). The input transducer (IT) may comprise a microphone (M) for converting sound to an electric signal. The input transducer may further comprise an analogue to digital converter (AD) for converting an analogue electric signal from the microphone (M) to a digitized signal (X) comprising a stream of digitized samples (cf. FIG. 1B). The input transducer (IT) may comprise further circuitry for processing the input signal, such as e.g. an analysis filter bank to provide the electric input signal in a time frequency representation (k,n) as the case may be (k, n being frequency and time-frame indices, respectively). The forward path further comprises a processor (PRO) for processing the electric input signal (X), or a signal derived therefrom (e.g. a feedback corrected signal E), and for providing a processed signal (U). The forward path further comprises an output transducer (OT) for converting the processed signal (U), or a signal derived therefrom, to stimuli perceivable by the user as sound (‘Acoustic output’). The forward path is configured to provide a forward path transfer function (F). The forward path transfer function (F) may e.g. be configured to compensate for a hearing impairment of a user of the hearing aid. The hearing aid (HA) further comprise a feedback control system (FBC) for handling external feedback from the output transducer (OT) to the input transducer (IT), cf. feedback sound signal v. The feedback control system comprises an adaptive filter (AF) comprising an algorithm part (ALG) and a variable filter part (Filter). The algorithm part (ALG) comprises an adaptive algorithm configured to provide updated filter coefficients (Ĥ) to the variable filter (FIL). The updated filter coefficients represent an estimate (Ĥ) of the current transfer function (H) of the feedback path (FBP). The adaptive filter (AF) is configured to provide an estimate ({circumflex over (V)}) of the current feedback signal (v (V)) from the output transducer (OT) to the input transducer (IT) in dependence of an error signal E (X−{circumflex over (V)}) and a reference signal (processed signal U), and a further signal (Ĥ′.sub.post) providing an instant feedback estimate (in certain situations when the feedback path changes fast). The feedback control system further comprises a combination unit (CU) located in the forward path and configured to subtract the current estimate ({circumflex over (V)}) of the feedback signal (v (V)) from the electric input signal (X), and to provide a feedback corrected signal (E=X−{circumflex over (V)}), termed the error signal. The processor (PRO) is configured to base its processing on the error signal (E).

(20) In the embodiment of FIG. 1A, the feedback control system (PBC) further comprises a feedback change estimator (FCE) configured to—at least in certain situations when the feedback path changes fast (e.g. when a feedback instability criterion is fulfilled)—provide an instant estimate (Ĥ.sub.post) of the feedback path transfer function in dependence of the forward path transfer function (F), and, optionally, of the current estimate (Ĥ.sub.pre) of the feedback path transfer function from the adaptive algorithm. The feedback control system further comprises an adaptive filter controller (AFC) for providing an update transfer function estimate (Ĥ′.sub.post) for the adaptive filter (AF) in dependence of the estimate (Ĥ.sub.post) of the instant feedback path transfer function. The estimate (Ĥ.sub.post) of the instant feedback path transfer function is intended to be provided from one time index (n) to the next (n+1) (as opposed to the current estimate (Ĥ.sub.pre) of the feedback path transfer function provided by the (adaptive algorithm (ALG) of the) adaptive filter (AF)). This is particularly relevant in case of a sudden change in the feedback path, where the adaptive estimate (Ĥ.sub.pre) will take some time instances to converge towards the changed feedback path (depending on the algorithm and the adaptation rate, e.g. on a time step of each iteration). The instant estimate (Ĥ′.sub.post) of the feedback path transfer function is intended to override the estimate (Ĥ.sub.pre) of the current feedback path transfer function provided by the adaptive filter (AF) to thereby provide a faster convergence of the adaptive algorithm (ALG). The feedback control system may comprise a feedback instability detector for monitoring the fulfillment of a feedback path instability criterion (e.g. indicating a sudden change or instability of the feedback path transfer function). The feedback instability detector may e.g. form part of or be connected to the feedback change estimator (FCE). It is the intention that the adaptive algorithm continues its feedback path estimation using the estimate (Ĥ′.sub.post) of the instant feedback path transfer function and to let the adaptive algorithm continue its adaptation from there (see e.g. FIG. 4). In such case (after a sudden change of the feedback path, e.g. upon fulfillment of the feedback instability criterion), the resulting estimate of the feedback path transfer function provided by the feedback control system (Ĥ) is equal to (Ĥ.sub.post(n) or Ĥ′.sub.post(n)), whereas under ‘stable’ (or slowly changing) feedback path conditions, the resulting estimate of the feedback path transfer function provided by the feedback control system (Ĥ) is equal to the current estimate (Ĥ.sub.pre(n)) of the feedback path transfer function provided by the adaptive algorithm.

(21) FIG. 1B shows a second embodiment of a hearing aid (HA) comprising a feedback control system (FBC) according to the present disclosure. The embodiment of FIG. 1B is similar to the embodiment of FIG. 1A. In the embodiment of FIG. 1B the input transducer (IT) is shown to comprise a microphone (M) for converting sound to an electric signal, and an analogue to digital converter (AD) for converting an analogue electric signal from the microphone (M) to a digitized signal (X) comprising a stream of digitized samples. As in FIG. 1A, the input transducer (IT) may comprise further circuitry for processing the input signal, such as e.g. an analysis filter bank to provide the electric input signal in a time frequency representation (k,n). Further, in the embodiment of FIG. 1B the output transducer (OT) is shown to comprise digital to analogue converter (DA) for converting a stream of digitized samples to an analogue signal which is fed to a loudspeaker (SPK) for converting the analogue signal to sound (‘Acoustic output’). The output transducer (OT) may alternatively comprise a vibrator of a bone conducting hearing aid. The output transducer (OT) may further comprise a synthesis filter bank for converting a frequency sub-band representation of the output signal to a time domain signal. Compared to the embodiment of FIG. 1A, the embodiment of FIG. 1B further comprises an open loop gain estimator (OLGE) for providing an instant open loop gain estimate ({circumflex over (L)}) of the forward path of the hearing aid. In the embodiment of FIG. 1B, the feedback change estimator (FCE) is configured to provide the instant estimate (Ĥ.sub.post) of the feedback path transfer function in dependence of the forward path transfer function (F) received from the processor (PRO), as well as the instant open loop gain estimate ({circumflex over (L)}) received from the open loop gain estimator (OLGE) and, optionally, in dependence of the current estimate (Ĥ.sub.pre) of the feedback path transfer function provided by the adaptive algorithm. The instant open loop gain estimate (E) of the open loop transfer function (termed {circumflex over (L)}.sub.fast(k,n)) may e.g. be provided as described in the following.

(22) In the following, a method to instantly estimate the open loop transfer function L(k,n) of a hearing aid is proposed. A corresponding fast estimate {circumflex over (L)}.sub.fast(k,n) is then used to detect instability and to improve the adaptive filter estimate Ĥ(k,n) of the feedback path transfer function upon critical changes of feedback situations. This is illustrated in the below flowchart of FIG. 2.

(23) FIG. 2 shows a flowchart describing a scheme for updating a feedback estimate of a feedback control system according to the present disclosure.

(24) Instant Open Loop Transfer Function Estimate. Compute Instability Measure (e.g., Magnitude, Phase, Derivatives etc.):

(25) First, in order to decide on stability or instability, we estimate a fast (or ‘instant’) open loop transfer function denoted by {circumflex over (L)}.sub.fast(k,n). The fast/instant open loop transfer function can he calculated in several ways, e.g. as

(26) ${\hat{L}}_{\mathrm{fast}}\left(k,n\right)=E\left(k,n\right)/E\left(k,n-D\right).Math.\begin{array}{c}\mathrm{if}.Math.{\hat{L}}_{\mathrm{fast}}\left(k,n\right).Math.<\mathrm{threshold}=>\mathrm{stable}\\ \mathrm{else}=>\mathrm{unstable}\end{array}$
where E(k,n) represents the so-called error signal in a typical adaptive filter configuration (see signal E in FIGS. 1A, 1B), D represents a loop delay (se e.g. FIG. 3) of the audio path of the hearing aid, so that E(k,n−D) represents the error signal one loop delay earlier than E(k,n). If there is an instant and critical change of feedback path from H.sub.pre(k,n) to H.sub.post(k,n) (see e.g. FIG. 4), we expect/assume the estimation of {circumflex over (L)}.sub.fast (k,n) is very accurate. Such a critical change of feedback path may lead to system instability. The reason that it can be assumed that the estimate of the instant open loop transfer function is quite accurate in this case is because the feedback to signal ratio (e.g. |V|/|S.sub.x| in FIGS. 1A 1B) is high as there is a significant portion of feedback signal (V) after such a change from H.sub.pre(k,n) to H.sub.post(k,n).
If Instability, Compute Instant Feedback Transfer Function:

(27) Then, if a critical change of the feedback path has been detected, the estimate {circumflex over (L)}.sub.fast(k,n) and the known forward path transfer function F(k,n) are used to further make an approximation of the feedback path estimate Ĥ.sub.post, with the following steps.

(28) Based on Eq. (1), we define the true instant open loop transfer function L.sub.post(k,n) using the true feedback path transfer function after the instant feedback path change H.sub.post(k,n), and the current feedback path estimate Ĥ.sub.post(k,n) from just before the instant path change, as
L.sub.post(k,n)=(H.sub.post(k,n)−Ĥ.sub.pre(k,n)).Math.F(k,n),  (2)
after rearranging, we obtain,
H.sub.post(k,n)=L.sub.post(k,n)/F(k,n)+Ĥ.sub.pre(k,n).  (3)

(29) Now, replacing the true L.sub.post(k,n) with the estimate {circumflex over (L)}.sub.fast_post(k,n), we can then make an approximation Ĥ.sub.post(k,n) of the unknown H.sub.post(k,n), as
Ĥ.sub.post(k,n)={circumflex over (L)}.sub.fast_post(k,n)/F(k,n)+Ĥ.sub.pre(k,n).  (4)
Update the Adaptive Filter Estimate:

(30) Finally, control parameters α,β=[0 . . . 1] may be introduced. The control parameters are intended for controlling an update of the adaptive filter estimate Ĥ(k,n) based on the estimate of instant feedback path Ĥ.sub.post(k,n) from Eq. (4), as
Ĥ′.sub.post(k,n)=α.Math.{circumflex over (L)}.sub.fast_post(k,n)/F(k,n)+β.Math.Ĥ.sub.pre(k,n).  (5)

(31) In an extreme exemplary case, where the magnitude of {circumflex over (L)}.sub.post(k,n) is big, e.g. ≥10 dB, and assume it is due to an instant change of H.sub.post(k,n) and |H.sub.post(k,n)|>>|Ĥ.sub.pre(k,n)|, i.e., the contribution from the adaptive filter Ĥ.sub.pre(k,n) is negligible compared to the instant feedback path change H.sub.post(k,n) right after the critical change of the feedback situation, we would update the feedback path estimate by using Ĥ(k,n) as
Ĥ(k,n)≈{circumflex over (L)}.sub.pst(k,n)/F(k,n),  (6)
by setting the parameters α=1 and β=0 in Eq. (5). In other less extreme cases, we would use the full equation in Eq. (5) with appropriate values of α and β. The parameters of α and β could be chosen based on the magnitude of the loop gain estimate {circumflex over (L)}.sub.post(k,n), e.g. if |{circumflex over (L)}.sub.post(k,n)| is high (e.g. ≥6 dB) α=1 and β=0; if e.g. |{circumflex over (L)}.sub.post(k,n)| is medium α=0.5 and β=0.5.

(32) This method/procedure is possible, since the estimation of {circumflex over (L)}.sub.fast(k,n) can be done very fast and reliably when the true magnitude of the open loop transfer function is indeed high, followed by a critical feedback path change; hence, the estimation of Ĥ.sub.post(k,n) in Eq. (4) is possible, and it is much faster than a traditional feedback cancellation system to reach Ĥ(k,n)=H.sub.post(k,n).

(33) Therefore, it makes sense to use Eq. (5) to make an instant update of the adaptive filter estimate Ĥ(k,n). The advantage of this is an increased convergence/tracking ability without sacrificing steady state error.

(34) FIG. 4 schematically illustrates an example of an adaptive algorithm which is extraordinarily updated (after a sudden change in the feedback path) at a given value of a time index (n*) using the Ĥ.sub.post(k,n*) value, and how the algorithm continues its convergence after the abrupt change. The (physical) feedback change may occur from one time index n* to the next (n*+1). Or the feedback change may occur over a number of subsequent time indices (i.e. over one or more units of the time index), One unit of the time index may e.g. be equal to the duration of a time frame (which e.g. if a time frame contains 64 time samples produced at a sampling rate of 20 kHz amounts to 3.2 ms). A sudden change of feedback path may e.g. occur over the order of up to 1 s.

(35) FIG. 3 shows the feedback loop of a hearing aid comprising an electric forward path from input to output transducer, and an acoustic (and/or mechanical) feedback loop from output to input transducer.

(36) Knowledge (e.g. an estimate or a measurement) of the length of one loop delay is assumed to be available (in advance or estimated during use).

(37) The loop delay D is defined as the time required for a signal travelling (once) through the acoustic loop, as illustrated in FIG. 3. The acoustic loop consists of the forward path (of the hearing aid), and the (acoustic) feedback path. The loop delay D is taken to include the processing delay d of the (electric) forward path (Forward Path (F)) of the hearing aid from input transducer (IT) to output transducer (OT) and the delay d′ of the acoustic feedback path (Feedback Path (H)) from the output transducer to the input transducer of the hearing aid, i.e. Loop Delay D=d+d′.

(38) Typically, the acoustic part d′ of the loop delay is much less than the electric (processing) part d of the loop delay, d′>>d (in particular when the forward path comprises processing of signals in frequency sub bands). The loop delay D may be approximated by the processing delay d of the forward path of the hearing aid (D≈d). The electric (processing) part d of the loop delay may e.g. be in the range between 2 ms and 10 ms, e.g. in the range between 5 ms and 8 ms, e.g. around 7 ms. The loop delay may be relatively constant over time (and e.g. determined in advance of operation of the hearing aid) or be different at different points in time, e.g. depending on the currently applied algorithms in the signal processing unit (d may e.g. be dynamically determined (estimated) during use). The hearing aid (HA) may e.g. comprise a memory unit wherein typical loop delays in different modes of operation of the hearing aid are stored. In an embodiment, the hearing aid is configured to measure a loop delay comprising a sum of a delay d of the forward path and a delay d′ of the feedback path. A predefined (or otherwise determined) test-signal may e.g. be inserted in the forward path, and its round trip travel time measured (or estimated), e.g. by identification of the test signal when it arrives in the forward path after a single propagation (or a known number of propagations) of the loop. The test signal may be configured to included significant content at frequencies where feedback is likely to occur (e.g. in a range between 1 and 5 kHz).

(39) FIG. 4 shows an exemplary time dependence of a true feedback path H and an estimated feedback path Ĥ according to the present disclosure. The graphs may represent values of a feedback path in a given frequency band (represented by frequency band index k (frequency domain)), or they may represent a full-band value (time domain). Values of (the magnitude of) feedback may e.g. be in the range between −200 dB and +10 dB, strongly dependent on the local acoustic environment around the hearing aid, e.g. within several in of the hearing aid (e.g. within a room wherein the hearing aid wearer is currently located). Each time unit, e.g. a time-frame length or a fraction thereof in case of overlapping time frames, may be of the order of 1 ms. For a given sampling frequency f.sub.s (e.g. 20 kHz) and a given number of samples N.sub.s per time frame (e.g. 64), the time frame length is N.sub.s/f.sub.s (e.g. 3.2 ms).

(40) FIG. 4 shows in solid tine an exemplary true feedback path transfer function H magnitude (Mag(H) ([dB]) versus time (Time, n [frame#]). The magnitude exhibits two sudden (abrupt) changes in an otherwise relatively stable course. The sudden changes in the true feedback path transfer function occurs at time instances n1 and n2. Such abrupt change may e.g. reflect that a telephone or other reflecting surface is held close to an ear of the user (as schematically indicated by the small insert drawings in FIG. 4 showing a telephone being moved to the ear and away from the car of the user at time instances n1 and n2, respectively). As indicated in the drawing on the time axis and in the solid curve (by the two crossing curved lines, ∫∫), there may be a time period between the two feedback path incidences (abrupt changes) that are not shown in the drawing (there may be a shorter (e.g. milli seconds) or longer (e.g. minutes) time period between n1 and n2, e.g. corresponding to a duration of a telephone conversation).

(41) FIG. 4 further shows by discrete solid dots the estimates (Ĥ.sub.pre) of the feedback path transfer function as provided by a prior art adaptive algorithm (e.g. the Least Mean Square (LMS) or the Normalized LMS (NLMS) algorithms) and a combination of a prior art algorithm and the modification proposed by the present disclosure.

(42) The lower left part of the dotted curve (before time n1, solid dots .circle-solid.) illustrates the estimate Ĥ of the true feedback path transfer function indicated by the solid curve by an adaptive feedback estimation algorithm according to the prior art (e.g. an LMS or an NLMS algorithm). In this time period (n<n1), the true feedback path is relatively stable and does not change faster than the adaptive algorithm can reasonably follow it (with a given adaptation rate or step size of the algorithm). At time instant n1, the true feedback path is abruptly changed because the user moves a telephone apparatus to the ear. This induces a change (ΔĤ(n1)) (increase) in the feedback path transfer function, which the adaptive algorithm cannot immediately follow, as indicated by the slowly increasing estimate indicated by the grey dots in FIG. 4 for time instances n1+1, etc. The value of the feedback path transfer function provided by the adaptive algorithm at time instance n1 prior to (or at) the sudden change of feedback path is denoted Ĥ.sub.pre(n1). To improve on the (erroneous) feedback estimate provided by the adaptive algorithm, the ‘next’ estimate of the feedback path transfer function provided by the adaptive algorithm is (forced to be) based on a corrected (estimated) true feedback path (after the sudden change). The value of the feedback path transfer function provided according to the present disclosure to the adaptive algorithm at time instance n1 after the sudden change of feedback path is denoted Ĥ.sub.post(n1) and indicated by the cross-hatched dot in FIG. 4. The value Ĥ.sub.post(n1) may e.g. be estimated as indicated above.

(43) The upper middle part: of the dotted curve (after time n1, but before time n2, solid dots .circle-solid.) illustrates the estimate Ĥ.sub.pre of the true feedback path transfer function indicated by the solid curve provided by an adaptive feedback estimation algorithm according to the prior art (starting from the value of the feedback path, Ĥ.sub.post(n1), estimated according to the present disclosure). In this time period (n2<n<n1) (again), the true feedback path is relatively stable and does not change faster than the adaptive algorithm can reasonably follow it (with the given adaptation rate or step size of the algorithm). At time instant n2, the true feedback path is abruptly changed because the user moves a telephone apparatus away from the ear. This induces a change (ΔĤ(n2)) (decrease) in the feedback path transfer function, which (again) the adaptive algorithm cannot immediately follow, as indicated by the slowly decreasing estimate indicated by the grey dots in FIG. 4 for time instances n2+1, etc. The value of the feedback path transfer function provided by the adaptive algorithm at time instance n2 prior to (or at) the sudden change of feedback path is denoted Ĥ.sub.pre(n2). To improve on the (erroneous) estimate of the feedback transfer function provided by the adaptive algorithm, the ‘next’ estimate of the feedback path provided by the adaptive algorithm is (forced to be) based on a corrected (estimated) true feedback path transfer function (after the sudden change). The value of the feedback path provided according to the present disclosure to the adaptive algorithm at time instance n2 after the sudden change of feedback path is denoted Ĥ.sub.post(n2) and indicated by the cross-hatched dot in FIG. 4. The value Ĥ.sub.post(n2) may e.g. be estimated as indicated above.

(44) The lower right part of the dotted curve (after time n2, solid dots .circle-solid.) illustrates the estimate Ĥ of the true feedback path transfer function indicated by the solid curve provided by an adaptive feedback estimation algorithm according to the prior art (starting from the value of the feedback path, Ĥ.sub.post(n2), estimated according to the present disclosure). In this time period (n>n2), the true feedback path is again relatively stable and does not change faster than the adaptive algorithm can reasonably follow it (with the given adaptation rate or step size of the algorithm).

(45) Thereby an improved adaptive algorithm can be provided.

(46) The output of the feedback estimation unit according to the present disclosure may (after a sudden change of the feedback path transfer function larger than a pre-determined threshold) be a value estimated according, to the present disclosure, and otherwise be a value provided by a prior art adaptive algorithm (e.g. an LMS or an NLMS algorithm with fixed or adaptively controlled step size/adaptation rate). The prior art adaptive algorithm may be configured to base its estimate a after a sudden change of the feedback path above a pre-determined threshold value on the value estimated according to the present disclosure.

(47) A Method of Detecting a Sudden Change in a Feedback/Echo Path:

(48) FIG. 5 shows a flow diagram for a method of detecting a sudden change of a feedback/echo path of a hearing aid or headset.

(49) The method may comprise at least some of the following steps: 1. Estimating a feedback path, e.g. using an adaptive algorithm. 2. Smoothing a gradient of the adaptive algorithm over time. 3. Perform an operation on the gradient, e.g. a logic operation, to provide a modified (smoothed) gradient. 4. Check whether the modified gradient fulfils an instability criterion, e.g. a threshold criterion. If the instability criterion is not fulfilled, repeat steps 1-4, otherwise go to step 5. 5. Determine a feedback path change from the gradient, and optionally 6. Update the adaptive feedback path estimate of the adaptive algorithm and/or adapt other processing of the device, e.g. directionality.

(50) The instability criterion may be fulfilled when the one or more gradient values or a combination (e.g. an average), such as a weighted combination (e. a weighted average) of the one or more gradient values are or is larger than a threshold value.

(51) FIG. 6 shows exemplary waveforms of signals from which the sudden change of feedbacklecho path can be identified according to a method of the present disclosure. The three (interrelated) waveforms of FIG. 6 illustrate time dependence of three different parameters during a time period of 0.2 s from t=0.4 s to t=0.6 s, cf. horizontal axis denoted Time [s] in the lower part of FIG. 6. A standard feedback cancellation system based on an adaptive filter estimation of the feedback/echo path is assumed to be active.

(52) The first (upper) plot, denoted ‘Feedback Path Change’, shows that there is a sudden or substantial (here ideally instantaneous) feedback path change at t=0.5 s. The size of the feedback path change is indicated along the vertical axis on a relative scale denoted ‘Change’ between 0 and 1.

(53) The second (middle) plot, denoted ‘Open Loop Magnitude’, shows open loop magnitude versus time in dependence of the feedback path change of the upper plot. The size of the open loop magnitude is indicated along the vertical axis denoted ‘Magnitude [dB]’ on a logarithmic scale between −20 dB and +20 dB. The sudden feedback change at t=0.5 results in a sudden change in open loop magnitude (of >20 dB) at t=0.5 s. It appears from the middle plot that it would take the adaptive filter more than 300 ms (from t=0.5 s to t≈0.53 s) before the open loop magnitude is again below the critical loop magnitude of 0 dB after the change (cf the crossing of the graph with the (bold) horizontal line representing 0 dB occurring at t≥0.53 s).

(54) The third (lower) plot, denoted Gradient Measure, shows the gradient versus time in dependence of the feedback path change of the upper plot. The size of the gradient measure is indicated along the vertical axis denoted ‘Magnitude []’ on a linear scale between 0 and 0.002. The lower plot illustrates that using the gradient method with a simple threshold, here e.g. TH ≈0.03 (cf. steps 1-3 in the method of FIG. 5). The detection of significant feedback/echo path change is already possible after a short time compared to a normal convergence time of several hundred ins (here after ˜5 ms).

(55) Thereby a correspondingly fast action can be taken, e.g. to make a change to the current value of the feedback estimate of the adaptive algorithm of the feedback cancellation system (cf. the (sudden) change from Ĥ.sub.pre to Ĥ.sub.post at time n1 in FIG. 4) and/or for influencing settings of a beamformer or performing other actions to the processing of the electric input signals.

(56) A Hearing Aid Comprising an In-Ear Microphone

(57) Hearing instruments for hearing loss compensation are currently programmed to a certain gain based on client data (hearing loss, age, gender, etc.) and a fitting rationale (e.g. NAL-NL2). The effective amplification at the tympanic membrane, however, can vary greatly based on the tolerances of the acoustic transducers (e.g. microphone and receiver loudspeaker)), the user's external ear anatomy, and the placement of the instrument on the ear. This variation may easily be of the order of 10-20 dB up to 4 kHz and even more at higher frequencies. Given the potential variations, adjusting the fitting rationale by a few dB will probably not have the desired audible effect for all potential users. Part of this variance can of course be compensated for by doing real ear measurements (REM) using probe microphone equipment, but these measurements are not performed for all fittings and do not compensate for the altered acoustical environment after removing and reinserting the instrument.

(58) One method to reduce this variance would be to place a monitor microphone inside the ear canal to measure the effective sound pressure level that is present at the ear drum. The amplification could then be measured and controlled accordingly in order to reach a defined target amplification. Previous attempts in adding a monitor microphone to an ITE instrument discovered many technical challenges. Furthermore, it requires an additional microphone which increases the size, the power consumption and complexity of the instrument. The present invention disclosure presents a hearing instrument setup, where at least some of these issues may be solved.

(59) Recent developments in feedback management promise that the goal of providing a feedback-free hearing instrument may not be far away. Removing the feedback constraint opens the door to new opportunities, such as a ‘reversed open invisible-in-canal (IIC)’ hearing aid. Reversed open IIC type of hearing aid according to the present disclosure may e.g. exhibit one or more of the following characteristics (cf. FIG. 7 for reference): The microphone (M) is placed in the housing (Housing) AFTER the receiver (SPK) in a direction towards (and close to) the tympanic membrane (Eardrum). The hearing instrument (HD) does not seal the ear canal (Ear canal) so that as much direct sound (S.sub.env) as possible reaches the tympanic membrane and the microphone (M), Both the receiver (SPK) and the microphone (M) are placed in the bony part (Bony part) of the ear canal and are hence protected against earwax. The setup may also be used as the external transducer unit of a receiver in the ear (RITE) type of hearing aid (where the microphone M may act as the microphone or one of the microphones of the RITE hearing aid).

(60) A further aspect of the present application relates to a hearing aid comprising of being constituted by an ITE-part adapted for being located in a bony part of the ear canal of a user (close to the eardrum). FIG. 7 shows a hearing aid according to this further aspect of the present disclosure when located in an ear canal close to the eardrum of a user.

(61) A hearing aid (HD) comprising an elongate housing configured to be located in a bony part of an ear canal of the user is furthermore provide by the present disclosure. The hearing aid comprises a forward path for processing an audio signal. The forward path comprises a) an input transducer for picking up sound in the ear canal and adapted to provide an electric input signal representing said sound, b) a signal processor for processing said electric input signal, or a signal originating therefrom, and providing a processed signal, and c) an output transducer configured to convert the processed signal to output sound in dependence of said electric input signal. The hearing aid may further comprise a feedback control system for estimating and cancelling, or reducing, signal components in said electric input signal originating from a feedback path from the output transducer to the input transducer and to provide a feedback corrected input signal. A cross-sectional area of said housing may be smaller than a cross-sectional are of said bony pan of the ear canal, when the hearing is mounted as intended. The input transducer and the output transducer may be mounted in the housing relative to each other so that the input transducer is closer to the eardrum than the output transducer.

(62) The housing may have a longitudinal direction in a direction towards the eardrum, when the hearing aid is mounted as intended. The cross-sectional area of the housing may be smaller than a cross-sectional are of the bony part of the ear canal along the longitudinal direction of the housing. Thereby it is achieved that sound can relatively freely pass from the environment to the eardrum around or along the housing of the hearing aid when mounted as intended in the ear canal of the user.

(63) The housing may comprise a sound outlet from the output transducer in a direction towards the eardrum when the hearing aid is mounted as intended in the ear canal of the user. Thereby sound vibrations from the output transducer are directed towards the eardrum of the user. The output transducer may be constituted by or comprise a loudspeaker.

(64) The housing may comprise a sound inlet to the input transducer in a direction towards the environment, when the hearing aid is mounted as intended in the ear canal of the user. Thereby sound vibrations from the environment (and possibly from the output transducer) are directed towards the eardrum of the user. The output transducer may be constituted by or comprise a microphone and or a vibration sensor, e.g. an accelerometer, or a bone conduction microphone.

(65) The hearing aid may comprise a user interface allowing remote control of functionality of the hearing aid, e.g. on/off, volume and program shift. The housing may comprise a wireless receiver forming part of the user interface.

(66) The hearing aid may comprise a battery (e.g. a rechargeable battery), or other energizing means, for powering the components enclosed in the housing. The battery (or other energizing means) may be located in the housing.

(67) As illustrated in FIG. 7, a hearing aid (e.g. or the ITE-part of the hearing aid) according to the present disclosure comprises a forward path for processing an audio signal. The forward path may comprise a) an input transducer (M, e.g. a microphone) for picking up sound (S.sub.env+S.sub.HD) in the ear canal (ear canal) and providing an electric input signal representing said sound, b) a signal processor (Amplifier) for processing (e.g. amplifying or attenuating) said electric input signal (or a signal originating therefrom) and providing a processed signal, and c) an output transducer (SPK, e.g. a loudspeaker) configured to convert the processed signal to output sound (S.sub.HD) in dependence of said electric input signal. The hearing aid (or the ITE-part of the hearing aid) further comprises a feedback control system for estimating and cancelling (or reducing) signal components in a signal of the forward path originating from a feedback path from the output transducer to the input transducer (cf. FIG. 8).

(68) The mechanical setup of the hearing instrument is of course very prone to feedback, so this invention is dependent on a feedback canceller (see FIGS. 7, 8): 1. The microphone signal y(n) represents the sound that the user experiences at the tympanic membrane (Eardrum). This signal is very useful to have access to, since it represents what the user potentially hears directly, including what amount of comb filter effect is present, what the final sound pressure level is at the tympanic membrane, and so on. That is why this microphone is also called a monitor microphone. 2. The microphone signal y(n) is the sum of two separate components, the direct sound (S.sub.env) represented by x(n) in FIG. 8 and the feedback sound (S.sub.HD) represented by signal from the receiver v(n) in FIG. 8. The feedback canceller (comprising feedback estimator FB.sub.est and sum unit ‘+’) subtracts the feedback estimate v.sub.est(n) from the electric input signal (microphone signal y(n)) such that only the direct sound x(n) remains (in the ideal case where v.sub.est(n)=v(n)), which is used as input to the signal processor (PRO) for applying one or more processing algorithms to the feedback corrected input signal e(n) (output of sum unit (‘+’). The one or more algorithm may e.g. include noise reduction, hearing loss compensation, etc. The level difference between this direct sound signal x(n) and the microphone signal y(n) may represent the effective amplification that the user receives. 3. The feedback canceller derives the feedback signal v.sub.est(n) from the hearing instrument output signal u(n) by applying the estimated feedback path h.sub.est(n) (impulse response, or transfer function). This estimated feedback path is, in the ideal case, equal to the feedback path h(n), which has two components: a. The transfer function of the receiver (SPK): Certain changes in h(n) could indicate, for example, a malfunctioning receiver. b. The in-ear transfer function from the receiver (SPK) to the microphone (M), which is defined by the acoustics of the ear canal: Certain changes in h(n) could e.g. indicate that the user occludes the ear canal with a finger. This can be used as a means to interact with the instrument (turn it off, change the program, etc.).

(69) As described in point 2 above, the effective amplification at the user's ear can estimated by calculating the level difference between the signals x(n) and y(n). And as both the direct sound x(n) and the amplified sound y(n) are measured with the same microphone, any hardware tolerance of that microphone is subtracted out when calculating the effective amplification.

(70) The difference between the effective amplification and a given target amplification can then determined in order to adjust the Hearing Instrument amplification f(n) accordingly and to finally converge to the target amplification.

(71) The hearing device that can thus accurately measure the effective amplification provided to the user. This information can be used in two different ways: 1) adjust the instrument gain during wearing time in order to converge to the desired target amplification, or 2) log the difference between the effective and the desired target amplification over time and provide a suggested gain adjustment to a Hearing Care Professional (HCP) or directly to the user.

(72) Further, the tympanic membrane signal may be used as input signal for the HI. The tympanic membrane signal can be captured by a monitor microphone as described above, but also other techniques might be used. Examples of such other techniques are: a laser vibrometer, capacitive sensors, a measurement device directly coupled to the tympanic membrane or the middle ear ossicles. The amplified sound is then also applied to the tympanic membrane by means of a traditional receiver (over the air) or by any other means like actuators mounted directly on the membrane or the ossicles.

(73) A difference of this idea over previous solutions is that it requires strong acoustic feedback in order to work. The direct sound and the amplified sound have to add up at the microphone so that we can estimate the effective amplification. In other words, the ear canal has to be as open as possible in contrast to other solutions where the canal is substantially sealed.

(74) Further, this monitoring works using only one single microphone. Other monitor microphone solutions have been proposed before, but the monitor microphone is normally used just for monitoring, not as the primary input source to the hearing aid.

(75) A hearing instrument with the described hardware characteristics has a number of other potential benefits. Truly invisible, i.e. it cannot be seen externally. No wind noise. Preservation of the natural cues from the Pinna AND the external ear canal resonance. When turned off, it is just as if you had no hearing aid on, i.e. it does not occlude the ear canal.

(76) It is intended that the structural features of the devices described above, either in the detailed description and/or in the claims, may be combined with steps of the method, when appropriately substituted by a corresponding process.

(77) As used, the singular forms “a,” “an,” and “the” are intended to include the plural forms as well (i.e. to have the meaning “at least one”), unless expressly stated otherwise. It will be further understood that the terms “includes,” “comprises,” “including,” and/or “comprising,” when used in this specification, specify the presence of stated features, integers, steps, operations, elements, and/or components, but do not preclude the presence or addition of one or more other features, integers, steps, operations, elements, components, and/or groups thereof. It will also be understood that when an element is referred to as being “connected” or “coupled” to another element, it can be directly connected or coupled to the other element but an intervening element may also be present, unless expressly stated otherwise. Furthermore, “connected” or “coupled” as used herein may include wirelessly connected or coupled. As used herein, the term “and/or” includes any and all combinations of one or more of the associated listed items. The steps of any disclosed method is not limited to the exact order stated herein, unless expressly stated otherwise.

(78) It should be appreciated that reference throughout this specification to “one embodiment” or “an embodiment” or “an aspect” or features included as “may” means that a particular feature, structure or characteristic described in connection with the embodiment is included in at least one embodiment of the disclosure. Furthermore, the particular features, structures or characteristics may be combined as suitable in one or more embodiments of the disclosure. The previous description is provided to enable any person skilled in the art to practice the various aspects described herein. Various modifications to these aspects will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other aspects.

(79) The claims are not intended to be limited to the aspects shown herein but are to be accorded the full scope consistent with the language of the claims, wherein reference to an element in the singular is not intended to mean “one and only one” unless specifically so stated, but rather “one or more.” Unless specifically stated otherwise, the term “some” refers to one or more.

(80) Accordingly, the scope should be judged in terms of the claims that follow.

REFERENCES

(81) EP3291581A2 (Oticon) Jul. 3, 2018