Partner microphone unit and a hearing system comprising a partner microphone unit

Abstract

A partner microphone unit comprising a) a multitude microphones for picking up a sound from the environment providing corresponding electric input signals, each comprising a target signal component and a noise signal component; b) a multi-input unit noise reduction system for providing an estimate Ŝ of the target sound s comprising the person's voice and comprising a multi-input beamformer filtering unit coupled to said input units and configured to determine filter weights for providing a beamformed signal, wherein signal components from other directions than a direction of the target signal source are attenuated, whereas signal components from the direction of the target signal source are left un-attenuated; c) antenna and transceiver circuitry for establishing an audio link to another device; and wherein the multi-input beamformer filtering unit comprises an adaptive beamformer is provided.

Claims

1. A partner microphone unit configured to pick up target sound from a target sound source, the target sound s comprising a voice of a person, the partner microphone unit comprising: a multitude of input units IU.sub.i, i=1, 2, . . . , M, M being larger than or equal to two, each input unit comprising a microphone for picking up a sound from the environment of the partner microphone unit and configured to provide corresponding electric input signals, each electric input signal comprising a target signal component and a noise signal component; a multi-input unit noise reduction system for providing an estimate Ŝ of the target sound s comprising the person's voice, the multi-input unit noise reduction system comprising a multi-input beamformer filtering unit operationally coupled to said multitude of input units IU.sub.i, i=1, . . . , M, and configured to determine filter weights for providing a beamformed signal, wherein signal components from other directions than a direction of the target signal source are attenuated, whereas signal components from the direction of the target signal source are left un-attenuated or are attenuated less relative to signal components from said other directions, wherein the multi-input unit noise reduction system is configured to adaptively estimate a current look vector d(k,m) of the beamformer filtering unit for a target signal originating from a target signal source located at a specific location relative to the person wearing the partner microphone unit, wherein the specific location relative to the person is the location of the person's mouth; and antenna and transceiver circuitry for establishing an audio link to another device, wherein the multi-input beamformer filtering unit comprises an adaptive beamformer.

2. A partner microphone unit according to claim 1 wherein the multi-input beamformer filtering unit comprises an MVDR beamformer.

3. A partner microphone unit according to claim 1 comprising a voice activity detector for estimating whether or not or with which probability a voice of the person is present in the current sound from the environment and providing a voice activity control signal indicative thereof, or is configured to receive such voice activity control signal from another device.

4. A partner microphone unit according to claim 3 wherein at least two of the input units comprise a level detector for detecting an input level of the sound picked up by the microphones of the input units in question, and wherein the voice activity detector is configured to base the voice activity control signal on the difference between the input levels of the respective electric input signals of the microphones.

5. A partner microphone unit according to claim 1 adapted to be worn by a person.

6. A partner microphone unit according to claim 1 wherein the multi-input unit noise reduction system is configured to estimate a noise power spectral density of disturbing background noise when the voice of the person is not present, or is present with probability below a predefined level, or to receive such estimate from another device.

7. A partner microphone unit according to claim 1 comprising a memory comprising a predefined reference look vector defining a reference spatial direction from the partner microphone unit to the target sound source.

8. A partner microphone unit according to claim 1 wherein the multi-input unit noise reduction system is configured to update a look vector when the target sound is present or present with a probability larger than a predefined value.

9. A partner microphone unit according to claim 8 configured to limit said update of the look vector by comparing currently determined beamformer weights corresponding to a current look vector with default weights corresponding to the reference look vector, and to constrain or neglect the currently determined beamformer weights if these differ from the default weights more than a predefined absolute ort relative amount.

10. A partner microphone unit according to claim 1 comprising a memory comprising predefined reference inter-microphone noise covariance matrices of the partner microphone unit.

11. A partner microphone unit according to claim 10 configured to control the update of the noise power spectral density of disturbing background noise by comparing currently determined inter-microphone noise covariance matrices with the reference inter-microphone noise covariance matrices, and to constrain or neglect the update of the noise power spectral density of disturbing background noise if the currently determined inter-microphone noise covariance matrices differ from the reference inter-microphone noise covariance matrices by more than a predefined absolute or relative amount.

12. A partner microphone unit according to claim 1 comprising an attachment element for attaching said microphone unit to the user.

13. A partner microphone unit according to claim 1 configured to transmit the estimate Ŝ of the target sound s comprising the person's voice to the other device, e.g. a hearing device.

14. A hearing system comprising a partner microphone unit according to claim 1 and a hearing device, e.g. a hearing aid, wherein the hearing device comprises antenna and transceiver circuitry for establishing a communication link to and receiving an audio signal comprising an estimate of the target sound s comprising a voice of the person from said partner microphone unit.

15. A hearing system according to claim 14 wherein the hearing device comprises an input transducer for picking up sound from the environment of the hearing device and providing an electric hearing device input signal, a signal processing unit for applying one or more processing algorithms to the electric hearing device input signal, or a signal originating therefrom, and providing a processed hearing device signal, and an output unit for providing stimuli perceived by a user as sound based on the processed hearing device signal or a signal originating therefrom, and an analysis unit configured to analyse the audio signal received from the partner microphone unit, and to generate one or more control signals for controlling said one or more processing algorithms.

16. A hearing system according to claim 14 wherein a forward path is defined in the hearing device from an input transducer to an output unit and wherein the forward path comprises a selection or mixing unit allowing the audio signal received from the partner microphone unit to be combined with a signal of the forward path or to be switched into the forward path instead of a signal picked up by the input transducer.

17. A hearing system according to claim 14 wherein the hearing device comprises a delay unit configured to delay the audio signal received from the partner microphone unit with a predefined or dynamically determined delay time.

18. A hearing system according to claim 14 wherein the hearing device comprises a control unit for receiving said estimate Ŝ of the target sound s comprising the person's voice from the partner microphone and configured to dynamically control transient reduction or maximum gain of the electric hearing device input signal or a signal originating therefrom.

19. A hearing system according to claim 14 comprising a multitude of partner microphone units.

20. Use of a partner microphone unit as claimed in claim 1 in a hearing aid system to pick up and reduce noise in a voice of a speaker or communication partner and to transmit the noise reduced signal to a hearing device, e.g. a hearing aid, worn by a user.

21. A partner microphone unit according to claim 1 wherein the multi-channel variable beamformer filtering unit comprises an MVDR filter providing filter weights w.sub.mvdr(k,m), said filter weights w.sub.mvdr(k,m) being based on a look vector d(k,m) and an inter-input unit covariance matrix for the noise signal, wherein the look vector d(k,m) is an M-dimensional vector comprising elements (i=1, 2, . . . , M), the i.sup.th element d.sub.i(k,m) defining an acoustic transfer function from the target signal source at a given location relative to the input units of the partner microphone unit to the i.sup.th input unit, or the relative acoustic transfer function from the i.sup.th input unit to a reference input unit.

22. A hearing system according to claim 14 wherein the hearing device comprises a hearing aid adapted to provide a frequency dependent gain, and/or a level dependent compression, and/or a transposition of one or more frequency ranges to one or more other frequency ranges, to compensate for a hearing impairment of a user.

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 exemplary use scenario of a hearing system according to the present disclosure comprising a partner microphone unit and a pair of hearing devices, and

(3) FIG. 1B shows a second exemplary use scenario of a hearing system according to the present disclosure comprising a partner microphone unit and a pair of hearing devices,

(4) FIG. 2 shows a block diagram of a multi-input beamformer-noise reduction system of a partner microphone unit according to the present disclosure,

(5) FIG. 3 shows an exemplary block diagram of an embodiment of a hearing system according to the present disclosure comprising a partner microphone unit and a hearing device,

(6) FIG. 4 illustrates a typical situation where an acoustically propagated target signal is received later than a wirelessly transmitted target signal at the hearing aid use, and

(7) FIG. 5 shows an exemplary block diagram of a hearing device wherein the wirelessly received signal is used for improved transient detection (for transient reduction) and level estimation (for compression and amplification).

(8) 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.

(9) 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

(10) 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.

(11) The electronic hardware may include microprocessors, microcontrollers, digital signal processors (DSPs), field programmable gate arrays (FPGAs), programmable logic devices (PLDs), gated logic, discrete hardware circuits, and other suitable hardware configured to perform the various functionality described throughout this disclosure. 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.

(12) FIGS. 1A and 1B shows two exemplary use scenarios of a hearing system according to the present disclosure comprising a partner microphone unit (PMIC) and a pair of (left and right) hearing devices (HD.sub.l, HD.sub.r). The left and right hearing devices (e.g. forming part of a binaural hearing aid system) are worn by a user (U) at left and right ears, respectively. The partner microphone is worn by a communication partner or a speaker (TLK), whom the user wishes to engage in discussion with and/or listen to. The partner microphone (PMIC) may be a unit worn by a person (TLK) that at a given time only intends to communicate with the user (U). In a particular scenario, the partner microphone (PMIC) may form part of a larger system (e.g. a public address system), where the speaker's voice is transmitted to the user and possible other users of hearing devices, and possibly acoustically broadcast via loudspeakers as well. The partner microphone according to the present disclosure may be used in either situation. The multi-input microphone system of the partner microphone is configured to focus on the target sound source (the voice of the wearer) and hence direct its sensitivity towards its wearer's mouth, cf. (ideally) cone-formed beam (BEAM) from the partner microphone unit to the mouth of the speaker (TLK). The target signal thus picked up is transmitted to the left and right hearing devices (HD.sub.l, HD.sub.r) worn by the user (U). FIG. 1A and FIG. 1B illustrate two possible scenarios of the transmission path from the partner microphone unit to the left and right hearing devices (HD.sub.l, HD.sub.r).

(13) FIG. 1A shows a hearing system comprising a partner microphone (PMIC), a pair of haring devices (HD.sub.l, HD.sub.r) and (intermediate) auxiliary device (AD). The solid arrows indicate the path of an audio signal (PS) containing the voice of the person (TLK) wearing the partner microphone unit from the partner microphone unit (PMIC) to the auxiliary device (AD) and on to the left and right hearing devices (HD.sub.l, HD.sub.r). The (intermediate) auxiliary device (AD) may be a mere relay station or may contain various functionality, e.g. provide a translation from one link protocol or technology to another (e.g. from a far-field transmission technology, e.g. based on Bluetooth to a near-field transmission technology (e.g. inductive), e.g. based on NFC or ZigBee or a proprietary protocol. Alternatively the two links may be based on the same transmission technology, e.g. Bluetooth or similar standardized or proprietary scheme.

(14) FIG. 1B shows a hearing system comprising a partner microphone (PMIC), and a pair of haring devices (HD.sub.l, HD.sub.r). The solid arrows indicate the direct path of an audio signal (PS) containing the voice of the person (TLK) wearing the partner microphone unit (PMIC) from the partner microphone unit to the left and right hearing devices (HD.sub.l, HD.sub.r). The hearing system is configured to allow an audio link to be established between the partner microphone unit (PMIC) and the left and right hearing devices (HD.sub.l, HD.sub.r). The partner microphone unit (PMIC) comprises antenna and transceiver circuitry to allow (at least) the transmission of audio signals (PS), and the left and right hearing devices (HD.sub.l, HD.sub.r) comprises antenna and transceiver circuitry to allow (at least) the reception of audio signals (PS) from the partner microphone unit (PMIC). This link may e.g. be based on far-field communication, e.g. according to a standardized (e.g. Bluetooth or Bluetooth Low Energy) or proprietary scheme.

(15) FIG. 2 shows a block diagram of a multi-input beamformer-noise reduction system (NRS) of a partner microphone unit (PMIC) according to the present disclosure.

(16) The solution in more detail involves building a dedicated beamformer+single-channel noise reduction (SC-NR) algorithm, similar to the so-called ‘MOE system’ proposed in [Kjems and Jensen, 2012], which in this situation is able to adapt to the particular problem of retrieving a partner-mic. users voice signal from the noisy microphone signals, and reject/suppress any other sound sources (which can be considered to be noise sources in this particular situation). FIG. 2 shows a conceptual diagram of such a system.

(17) The multi-input noise reduction system may comprise a fixed beameformer with beam directed at an average person's mouth, when the partner microphone is positioned in a predefined position, e.g. on the chest of the person. In a preferred embodiment, an adaptive beamformer—single-channel noise reduction (SC-NR) system is provided in the partner microphone unit.

(18) The beamformer is adaptive in two ways: Firstly, when the partner-microphone wearer is silent, as e.g. detected by a VAD algorithm in the partner-microphone system, or in the hearing device, or another device, cf. optional connection via antenna and transceiver circuitry indicated in FIG. 2 by symbol ANT, e.g. based on voice activity from the far-end speaker, which is easily detected in the microphone unit (or in the hearing device or in the telephone). In such situation, inter-microphone noise covariance matrices may be updated to adapt the shape of the beampattern to allow for maximum spatial noise reduction. Secondly, when the partner-microphone wearer speaks, the beamformers' spatial direction (technically, represented by the so-called look-vector), is updated; this adaptation compensates for variation in partner-microphone position (across time and from wearer to wearer) and for differences in physical characteristics (e.g., head and shoulder characteristics) of the wearer of the partner microphone. Beamformer designs exist which are independent of the exact microphone locations, in the sense that they aim at retrieving the target signal in a minimum mean-square sense or in a minimum-variance distortion less response sense independent of the microphone geometry. In other words, the beamformer “does the best job possible” for any microphone configuration, but some microphone locations are obviously better than other.

(19) Furthermore, the SC-NR system, which may (or may not) be present, is adaptive to the level of the residual noise in the beamformer output; for acoustic situations, where the beamformer already rejected much of the ambient noise, the SNR in the beamformer output is already significantly improved, and the SC-NR system may be essentially transparent. However, in other situations, where a significant amount of residual noise is present in the beamformer output, the SC-NR system may suppress time-frequency regions of the signal, where the SNR is low, to improve the quality of the voice signal to be transmitted to a user of hearing device(s).

(20) The VAD algorithm may use the advantage that the partner microphone is located close to the target talker. If the microphone array is pointing towards the talker's mouth, the sound intensity level will be highest at the microphone closest to the mouth. This level difference may be used to determine when the talker is active.

(21) Before use, default beamformer weights are determined in an offline calibration process conducted in a sound studio with a head-and-torso-simulator (HATS, Head and Torso Simulator 4128C from Brüel & Kjær Sound & Vibration Measurement A/S) with play-back of voice signals from the dummy head's mouth, and a clip mounted in a default position on the “chest” of the dummy head. In this way, e.g., optimal minimum-variance distortion-less response (MVDR) beamformer weights may be found, which are hardwired, i.e. stored in a memory of the partner microphone unit.

(22) The adaptive beamformer—single-channel noise reduction (SC-NR) system allows a departure from the default beamformer weights, to take into account differences between the actual situation (with a real human user in a real (not acoustically ideal) room and a potentially with casual position of the microphone unit relative to the user's mouth) and the default situation (with the dummy in the sound studio and an ideally positioned partner microphone unit).

(23) The adaptation process may be supervised by comparing the adapted beamformer weights with the default weights, and potentially constrain the adapted beamformer weights (or fully dispense with the currently determined beamformer weights) if these differ too much from the default weights.

(24) The noise-reduced voice signal of the partner-mic. wearer is transmitted wirelessly to the hearing aid user, see FIGS. 3 and 4 below.

(25) FIG. 3 shows an exemplary block diagram of an embodiment of a hearing system according to the present disclosure comprising a partner microphone unit and a hearing device.

(26) FIG. 3 shows an exemplary block diagram of an embodiment of a hearing system according to the present disclosure comprising a partner microphone unit and a hearing device. FIG. 3 shows a hearing system comprising a hearing device (HD) adapted for being located at or in an ear of a user, or adapted for being fully or partially implanted in the head of the user, and a separate partner microphone unit (PMIC) adapted for being located at a person other than the user of the hearing devices and picking up a voice of the person. The partner microphone unit (PMIC) comprises a multitude M of input units IU.sub.i, i=1, 2, . . . , M, each being configured for picking up or receiving a signal x.sub.i (i=1, 2, . . . , M) representative of a sound PSP’ from the environment of the partner microphone unit (ideally from the person TLK, cf. reference From TLK in FIG. 3) and configured to provide corresponding electric input signals X.sub.i in a time-frequency representation in a number of frequency bands and a number of time instances. M is larger than or equal to two. In the embodiment of FIG. 3, input units IU.sub.1 and IU.sub.M are shown to comprise respective input transducers IT.sub.1 and IT.sub.M (e.g. microphones) for converting input sound x.sub.1 and x.sub.M to respective (e.g. digitized) electric input signals x′.sub.1 and x′.sub.M and each their filterbanks (AFB) for converting electric (time-domain) input signals x′.sub.1 and x′.sub.M to respective electric input signals X.sub.1 and X.sub.M in a time-frequency representation (k,m). All M input units may be identical to IU.sub.1 and IU.sub.M or may be individualized, e.g. to comprise individual normalization or equalization filters and/or wired or wireless transceivers. In an embodiment, one or more of the input units comprises a wired or wireless transceiver configured to receive an audio signal from another device, allowing to provide inputs from input transducers spatially separated from the partner microphone unit. The time-frequency domain input signals (X.sub.i, i=1, 2, . . . , M) are fed to a control unit (CONT) and to a multi-input unit noise reduction system (NRS) for providing an estimate Ŝ of a target signal s comprising the user's voice. The multi-input unit noise reduction system (NRS) comprises a multi-input beamformer filtering unit (BF) operationally coupled to said multitude of input units IU.sub.i, i=1, . . . , M, and configured to determine filter weights w(k,m) for providing a beamformed signal Y, wherein signal components from other directions than a direction of a target signal source (the partner person's voice) are attenuated, whereas signal components from the direction of the target signal source are left un-attenuated or are attenuated less relative to signal components from other directions. The multi-channel noise reduction system (NRS) of the embodiment of FIG. 3 further comprises a single channel noise reduction unit (SC-NR) operationally coupled to the beamformer filtering unit (BF) and configured for reducing residual noise in the beamformed signal Y and providing the estimate Ŝ of the target signal (the partner person's voice). The partner microphone unit may further comprise a signal processing unit (SPU) for further processing the estimate Ŝ of the target signal and provide a further processed signal pŜ. The partner microphone unit further comprises antenna and transceiver circuitry ANT, RF-Rx/Tx) for transmitting said estimate Ŝ (or further processed signal pŜ) of the partner microphone user's voice to another device, e.g. a hearing device (her indicated by reference ‘to HD, essentially comprising signal PSP, ‘partner speech’).

(27) The partner microphone unit (PMIC) further comprises a control unit (CONT) configured to provide that the multi-input beamformer filtering unit is adaptive. The control unit (CONT) comprises a memory (MEM) storing reference values of a look vector (d) of the beamformer (and possibly also reference values of the noise-covariance matrices C.sub.w(k)). The control unit (CONT) further comprises a voice activity detector (VAD) and/or is adapted to receive information (estimates) about current voice activity of the user of the partner microphone unit. Voice activity information is used to control the timing of the update of the noise reduction system and hence to provide adaptivity.

(28) The hearing device (HD) comprises an input transducer, e.g. microphone (MIC), for converting an input sound to an electric input signal INm. The hearing device may comprise a directional microphone system (e.g. a multi-input beamformer and noise reduction system as discussed in connection with the partner microphone unit, not shown in the embodiment of FIG. 3) adapted to enhance a target acoustic source in the user's environment among a multitude of acoustic sources in the local environment of the user wearing the hearing device (HD), e.g. the partner's voice. However—in a specific partner mode of operation—the hearing device microphone may be disabled or attenuated, so that the signal presented to the user is dominated by the signal comprising the voice of the partner as received from the partner microphone. The hearing device (HD) further comprises an antenna (ANT) and transceiver circuitry (Rx/Tx) for wirelessly receiving a direct electric input signal from another device, e.g. a communication device, or as here from the partner microphone unit, as indicated by reference ‘From PM/C’ and signal PSP (partner-speech) referring to the scenarios of FIGS. 1A and 1B. The transceiver circuitry comprises appropriate demodulation circuitry for demodulating the received direct electric input to provide the direct electric input signal INw representing an audio signal (and/or a control signal). The hearing device (HD) further comprises a selection and/or mixing unit (SEL-MIX) allowing to select one of the electric input signals (INw, INm) or to provide an appropriate (e.g. weighted) mixture as a resulting input signal RIN. The selection and/or mixing unit (SEL-MIX) is controlled by detection and control unit (DET) via signal MOD determining a mode of operation of the hearing device (in particular controlling the SEL-MIX-unit). The detection and control unit (DET), may e.g. comprise a detector for identifying the mode of operation (e.g. for detecting that the user is engaged in a conversation with or listening to a particular person wearing a partner microphone unit) or is configured to receive such information, e.g. from an external sensor and/or from a user interface.

(29) The hearing device comprises a signal processing unit (SPU) for processing the resulting input signal RIN and is e.g. adapted to provide a frequency dependent gain and/or a level dependent compression and/or a transposition (with or without frequency compression) of one or frequency ranges to one or more other frequency ranges, e.g. to compensate for a hearing impairment of a user. The signal processing unit (SPU) provides a processed signal PRS. The hearing device further comprises an output unit for providing a stimulus OUT configured to be perceived by the user as an acoustic signal based on a processed electric signal PRS. In the embodiment of FIG. 3, the output transducer comprises a loudspeaker (SP) for providing the stimulus OUT as an acoustic signal to the user (here indicated by reference ‘to U’ and signal PSP’ (partner-speech) referring to the scenarios of FIGS. 1A and 1B. The hearing device may alternatively or additionally comprise a number of electrodes of a cochlear implant or a vibrator of a bone conducting hearing device.

(30) The embodiment of FIG. 3 may e.g. exemplify the scenario of FIG. 1B.

(31) FIG. 4 shows a typical situation where an acoustically propagated target signal is received later than a wirelessly transmitted target signal at the hearing aid user.

(32) The transmission delay (depending on specific technology choices) can be as low as 3-5 ms. This delay is generally lower than the time it takes for the acoustic voice signal from the partner-microphone wearer to reach the microphones of the hearing aid user (the time for a sound wave to travel a meter is approximately 3 ms). So, for example, if the partner-microphone wearer is at a distance of 8 meters (and we assume a transmission delay of 5 ms), the wireless signal arrives at the hearing aids 8*3−5=19 ms earlier than the acoustic signal. In this situation, the wirelessly received signal is delayed by 19 ms, before it is mixed into the acoustically received signal (see Delay block and Mix block in FIG. 5).

(33) The fact that the wireless signal is generally received at the hearing aid several milliseconds before it needs to be played back to the hearing aid user, offers advantages for the signal processing blocks in the hearing aid, see FIG. 5 below for an exemplary block diagram.

(34) FIG. 5 shows an example block diagram of a hearing device receiving a target signal via a wireless link as well as via an acoustic propagation path. The wirelessly received signal is e.g. used for improved transient detection (block Transient Reduction) and level estimation (block Compression and amplification).

(35) For example, knowing part of the future signal (relative to the playback time) allows improved transient reduction (one can actually wait and see if an abrupt increase in signal energy is followed by an abrupt decrease before one has to decide whether a transient is present or not). Furthermore, the hearing-loss compensation (HLC) block in any hearing aid applies a frequency-dependent and time-varying gain to the input sound signal. The HLC gain in a particular frequency-band is a function of the signal power in the frequency-band at the playback time. As for the transient reduction situation, having “future” signal regions available (block Analysis of “future” wireless signal) allows a more accurate estimation of the signal power in a particular frequency region, which in turn allows a more accurate estimate of the HLC gain to be applied.

(36) Having more than one set of partner microphones or several microphones at different locations surrounding the listener will increase the probability of receiving a “future” wireless signal, hereby enabling the possibility of doing “acausal” processing.

(37) 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.

(38) 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 elements 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.

(39) 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.

(40) The claims are not intended to be limited to the aspects shown herein, but is 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.

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

REFERENCES

(42) [Kjems and Jensen; 2012] U. Kjems, J. Jensen, “Maximum likelihood based noise covariance matrix estimation for multi-microphone speech enhancement”, 20th European Signal Processing Conference (EUSIPCO 2012), pp. 295-299, 2012.