HEARING DEVICE COMPRISING AN ADAPTIVE FILTER BANK

Abstract

A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h(n), where m=0, 1, . . . , M−1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.

Claims

1. A hearing device, e.g. a hearing aid or a headset, comprising at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h(n), where m=0, 1, . . . , M−1 is a frequency band index, and n is a time index, a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one analysis filter bank in dependence of said current acoustic environment.

2. A hearing device according to claim 1 comprising a sound scene classifier configured to classify said acoustic environment into a number of different sound scene classes, and to provide a current sound scene class in dependence of a current representation, e.g. extracted features, of said at least one electric input signal.

3. A hearing device according to claim 2 wherein the sound scene classifier comprises a neural network.

4. A hearing device according to claim 2 wherein the sound scene classifier receives the at least one electric input signal as input.

5. A hearing device according to claim 2 wherein the sound scene classifier receives frequency domain input features, or a combination of time and frequency domain input features, extracted from said at least one electric input signal.

6. A hearing device according to claim 2 comprising a user interface allowing a user to influence functionality of the hearing device, including to allow the user to indicate the current acoustic environment, or by selection of a specific program, wherein each selected acoustic environment or program is associated with a specific prototype filter.

7. A hearing device according to claim 2 wherein the controller is configured to provide that a fading from one prototype filter to another is initiated when said sound scene classifier or said user changes its classification of the current acoustic environment from one sound scene class to another, such that the fading between the two filter banks maintain the same phase response.

8. A hearing device according to claim 2 wherein fading from one prototype filter to another is provided under the constraint that the two prototype filters have the same group delay.

9. A hearing device according to claim 8 wherein a fading time is greater than 1 second, such as greater than 5 seconds, or greater than 10 seconds.

10. A hearing device according to claim 1 wherein the different prototype filters are configured to exhibit the same group delay.

11. A hearing device according to claim 1 comprising at least two input transducers configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least two input transducers providing at least two electric input signals representative of said sound, a beamformer configured to provide beamformed signal in dependence of said at least two electric input signals and predefined and/or adaptively updated beamformer weights, wherein the beamformer weights are adapted in dependence of the selected prototype filter.

12. A hearing device according to claim 1 comprising an adaptive feedback control system comprising an adaptive algorithm for estimating a feedback path from said output transducer to said at least one input transducer, and wherein the hearing device is configured to control the adaptation rate of the adaptive algorithm in dependence of a change of the current acoustic environment.

13. A hearing device according to claim 12 wherein an adaptation rate of the feedback control system is temporarily increased when the different first prototype filter is applied in the analysis filter bank.

14. A hearing device according to claim 1 configured to provide that at least one of said prototype filters is dependent on a hearing loss of the user.

15. A hearing device according to claim 2 configured to provide that a specific sound scene is dependent on a measured sound level, a measured signal-to-noise ratio, a measured speech intelligibility estimate, a measured sound quality estimate, or a combination thereof.

16. A hearing device according to claim 1 adapted for being located at or in an ear of a user, or for being at least partially implanted in the head at an ear of the user.

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

18. A binaural hearing system comprising first and second hearing devices according to claim 1 wherein the hearing system is configured to change the prototype filters of the first and second hearing aids of the binaural hearing aid system simultaneously.

19. A binaural hearing system according to claim 18 wherein the prototype filters are adapted on each of the first and second hearing devices separately.

20. A method of operating a hearing device, e.g. a hearing aid, adapted for being located at or in an ear of a user, or for being at least partially implanted in the head at an ear of the user, the method comprising providing at least one electric input signal representative of sound from an acoustic environment around the user when the user is wearing the hearing device, providing said at least one electric input signal as a multitude of frequency sub-band signals, using a plurality of M first filters h.sub.m(n), where m=0, 1, . . . , M−1 is a frequency band index, and whose impulse responses are modulated from a first prototype filter h(n), n being a time index, processing said at least one electric input signal, or a signal originating therefrom, and providing a processed signal, providing stimuli perceivable as sound to the user in dependence of said processed signal, and wherein the step of providing said at least one electric input signal as a multitude of frequency sub-band signals comprises applying a different first prototype filter in dependence of said current acoustic environment.

Description

BRIEF DESCRIPTION OF DRAWINGS

[0103] 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:

[0104] FIG. 1 shows a frequency response of two different prototype filters exemplified by a Hamming prototype filter and a rectangular prototype filter,

[0105] FIG. 2 shows a filter bank with the prototype filter replicated at each centre frequency,

[0106] FIG. 3A shows a first example of an implementation of an analysis filter bank and a synthesis filter bank having an analysis prototype and a synthesis prototype filter; and

[0107] FIG. 3B shows a second example of an implementation of an analysis filter bank and a synthesis filter bank having an analysis prototype and a synthesis prototype filter,

[0108] FIG. 4A shows a first exemplary implementation of a hearing instrument with adaptive prototype analysis and synthesis filters; and

[0109] FIG. 4B shows a second exemplary implementation of a hearing instrument with adaptive prototype analysis and synthesis filters,

[0110] FIG. 5 shows a third exemplary hearing aid according to the present disclosure,

[0111] FIG. 6 shows a fourth exemplary hearing aid according to the present disclosure,

[0112] FIG. 7 shows a fourth exemplary hearing aid according to the present disclosure comprising beamformer, and

[0113] FIG. 8A shows a hearing system comprising a hearing aid and an auxiliary device in communication with each other, and

[0114] FIG. 8B shows the auxiliary device of FIG. 8A configured to implement a user interface for the hearing aid by running an application program from which a mode of operation of the hearing aid can be selected.

[0115] 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.

[0116] 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

[0117] 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.

[0118] 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.

[0119] The present application relates to the field of hearing aids or headsets, in particular to detection of a current acoustic environment around the hearing aid. The present disclosure proposes to change a prototype filter of a filter bank in dependence of a detected acoustic scene.

[0120] In hearing aids (or other audio processing devices, e.g. headsets), audio signals are typically divided into frequency channels in order to allow frequency dependent processing, such as hearing loss compensation, and/or beamforming-noise reduction. Hearing loss compensation may e.g. require significant gain differences between high and low frequencies. And a high stop-band attenuation is thus required in the filter bank. A prototype filter with a high stop-band attenuation will however typically have a quite broad main lobe resulting in more overlap between the neighbouring frequency bands. This is illustrated in FIG. 1 and FIG. 2. FIGS. 3A and 3B show respective examples of an implementation of an analysis filter bank (and synthesis filter bank) using prototype filters in a signal processing chain consisting of an analysis and a synthesis filter bank. It is important to notice that e.g. hearing aid applications have a limited amount of allowed latency (e.g. <10 ms) for processing an audio signal in a forward path from the microphone through the analysis and the synthesis filter banks before presenting the processed signal to the listener (e.g. via an output transducer). It is thus not an attractive option to increase the processing delay in order to achieve both high frequency resolution and a high stop-band attenuation at the same time.

[0121] Today, prototype filters are selected in order to fulfil a specific purpose. Hereby the prototype filter may be less optimal for other purposes. In the present disclosure, it is proposed to solve the problem by having an adaptive prototype filter which can change the prototype filter based on the specific situation, e.g. the acoustic environment.

[0122] FIG. 1 shows a frequency response of two different prototype filters exemplified by a Hamming prototype filter and a rectangular prototype filter. The magnitude response illustrates the amplification (0 to −50 dB) of the prototype filter around a centre frequency of the filter (located at frequency band 16 on the horizontal (frequency band) axis (here 32 bands in total)). It can be noticed that the Hamming prototype filter (solid line graph) has a much higher stop band attenuation (i.e. attenuation at other frequencies than the main lobe) compared to the rectangular prototype filter (dashed line graph). On the other hand, the rectangular prototype has a narrower main lobe, allowing better attenuation in the neighbouring frequency bands.

[0123] FIG. 2 shows a filter bank with the prototype filter replicated at each centre frequency of the available frequency channels. The vertical axis shows amplification between 0 and −50 dB. The horizontal axis is a frequency axis showing 32 uniform frequency channels. It can be seen that the Hamming window (top graph) has a higher stop-band attenuation compared to a rectangular prototype filter. On the other hand, the main lobe of the Hamming prototype filter is broader compared to the main lobe of the rectangular filter. The rectangular filter (bottom graph) will thus be more efficient at altering the gain of a single band where the effects of the neighbouring bands are less noticeable.

[0124] FIG. 3A shows a first example of an implementation of an analysis filter bank and a synthesis filter bank having an analysis prototype and a synthesis prototype filter. FIG. 3A shows a general illustration of a filter bank comprising an analysis filter bank (left part of FIG. 3A) and a synthesis filter bank (right part of FIG. 3A) (and an optional processing unit therebetween). The analysis filter bank comprises M filters H.sub.m(z) (h.sub.m(n) in the time domain, n), m=0, 1, . . . , M−1, which converts a time variant input signal X(z) comprising an audio signal into M time variant frequency band signals, each of which are (optionally) down-sampled with a down-sampling rate D to provide M time variant frequency band signals X.sub.m(z), m=0, 1, . . . , M−1, each representing a sub-band of the total frequency range of the input signal X(z). The M filters H.sub.m(z) of the analysis filter bank are generated from a (first) prototype filter h(n), by modulating the input signal

[00001]$h_m\left(n\right)=h\left(n\right)e^{\frac{j2\pi m\left(n-{\tau }_h\right)}{M}}$

[0125] where h is the prototype filter and τ.sub.h is the prototype filter delay of h.

[0126] Between the output of the analysis filter bank and the inputs of the synthesis filter bank, a signal processing unit (PRO) is shown (not forming part of the filter bank). The signal processing unit (PRO) may be configured to process the M time variant frequency band signals X.sub.m(z) and provide M processed time variant frequency band signals Y.sub.m(z) (e.g. to apply one or more (frequency dependent) signal processing algorithms to the input signals X.sub.m(z) (or signals derived therefrom). The synthesis filter bank (optionally) comprises M up-sampling units for up-sampling (processed) time variant frequency band signals Y.sub.m(z), m=0, 1, . . . , M−1, and M filters G.sub.m(z) (or g.sub.m(n) in the time domain) for converting the M up-sampled (processed) time variant frequency band signals to M (processed) filtered signals and a (delay-and-)sum unit (+) for providing a resulting time variant output signal Y(z) representing the audio signal or a processed version thereof (e.g. having been subject to spatial filtering and/or level and/or frequency dependent shaping, e.g. to compensate for a hearing impairment of a user). The M filters G.sub.m(z) of the synthesis filter bank are generated from a (second) prototype filter g(n):

[00002]$g_m\left(n\right)\propto g\left(n\right)e^{\frac{j2\pi m\left(n-{\tau }_g\right)}{M}}$

[0127] where g is the prototype filter and τ.sub.g is the prototype filter delay of g.

[0128] It is noted that the above is one way of transforming the signal to the Fourier domain. Other methods exist. More details may be found in the literature, e.g. in textbook [Vaidyanathan; 1993].

[0129] FIG. 3B shows a second example of an implementation of an analysis filter bank comprising an analysis prototype and a synthesis prototype filter. The analysis filter bank is represented by the left part of FIG. 3B including delay elements ‘z.sup.−1’, ‘Prototype window’ and ‘FFT’ unit. The analysis filter bank is configured to a time variant input signal x(n) comprising an audio signal into M time variant frequency band signals, x.sub.0(n), . . . , x.sub.M-1(n) (i.e. x.sub.m(n), m=0, 1, . . . , M−1). The synthesis filter bank is represented by the right part of FIG. 3B including ‘IFFT’ unit, ‘inverse window’, and sum and delay elements ‘+’ and ‘z’). The synthesis filter bank is configured to convert M time variant frequency band signals x.sub.m(n), m=0, 1, . . . , M−1, to a resulting time variant output signal y(n) representing the audio signal. The prototype filters of the analysis filter bank form part of the ‘Prototype window’. The prototype filters of the synthesis filter bank form part of the ‘inverse window’. In the analysis filter bank, the signals from the prototype filters converted to the time-frequency domain by a Fast Fourier transform algorithm (FFT), e.g. Discrete Fourier Transform (DFT) algorithm, or a Short Time Fourier Transform (STFT) algorithm, etc. (e.g. using matrix-multiplication with a modulation sequence). The matrix multiplications can in turn be implemented efficiently using a mapping procedure and the Fast Fourier Transform. Appropriate processing of the audio signal in the time-frequency domain (x.sub.m(n), m=0, 1, . . . , M−1) (cf. FIG. 3A) may be applied between the analysis and synthesis filter banks.

[0130] The number of frequency bands M may e.g. be larger than or equal to three, e.g. larger than or equal to 16.

[0131] Consider two situations for a hearing impaired person: [0132] 1) A quiet situation with a single talker. In this situation it is important to ensure audibility by applying a frequency-dependent amplification to the microphone signal. As the amplification scheme typically involves compression, low-intensity (softer) sounds will be amplified more than louder sounds. [0133] 2) A noisy situation with a single talker of interest. In order to ensure audibility of the signal of interest, the signal of interest not only has to be amplified, the background noise should as well be attenuated. As the sound environment is (typically) louder in such situation, typically less amplification is needed.

[0134] In the first case, a filter bank with a prototype filter having a high amount of stop-band attenuation is required. In the second case, a filter bank capable of extracting parts of a speech signal in a frequency channel with noisy neighboring channels is needed. We thus have two different auditory scenes, which could benefit from different types of prototype filters.

[0135] FIGS. 4A and 4B both illustrate a hearing instrument comprising a forward audio processing path for processing the audio signal in a time-frequency domain. The forward path from input transducer (IT), here comprising a microphone, to output transducer (OT), here comprising a loudspeaker, comprises a filter bank (FBA, FBS) comprising an adaptive analysis and synthesis prototype filter according to the present disclosure, and a signal processing unit (PRO), located between the analysis and synthesis filter banks. The signal processing unit (PR) is configured to apply one or more signal processing algorithms to an audio signal (X) of the forward path in a time-frequency domain and to provide a processed signal (Y). The input transducer (IT) is configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, and to provide an electric (time domain) input signal (x) representative of said environment sound. The analysis filter bank (FBA) is configured to convert the time domain electric input signal (x) comprising an audio signal into a multitude of time variant frequency band signals (X). The synthesis filter bank is configured to convert the multitude of time variant (processed) frequency band signals (Y) to a time domain electric input signal (y) comprising a processed version of the audio signal. The output transducer (OT) is configured to provide stimuli perceivable as sound to the user in dependence of said processed signal (y).

[0136] The hearing instrument (HD) further comprises a sound scene classifier (SA) either working in the time domain, as shown in FIG. 4A, or, as shown in FIG. 4B, in the (time-)frequency domain). A sound scene may be defined with a particular value or range of values of a (e.g. frequency dependent) parameter of the electric input signal (x, X). A parameter of the electric input signal may comprise (or be derived from) its level. A parameter of the electric input signal may comprise (or be derived from) its signal-to-noise-ratio. A parameter of the electric input signal may comprise (or be derived from) its sound quality estimate, e.g. its signal-to-noise-ratio. A parameter of the electric input signal may comprise (or be derived from) an estimate of its speech intelligibility.

[0137] Sound scenes may e.g. be interpreted as different based on (or influenced by) [0138] two measured (or estimated) levels, [0139] two measured (or estimated) sound quality estimates, e.g. signal-to-noise-ratios, or [0140] two measured (or estimated) speech intelligibility estimates

[0141] of the electric input signal.

[0142] The adaptation between different prototype filters may depend on a specific sound scene, e.g. a measured (or estimated) level, a measured (or estimated) SNR, a measured (or estimated) speech intelligibility estimate, a measured (or estimated) sound quality estimate, or a combination of the different parameters, e.g. expressed by a criterion containing one or more of said parameters.

[0143] A sound scene may also be estimated based on a (trained) sound scene classifier, e.g. as labeled sound scenes like traffic, babble, quiet, single talker, own voice, etc. The sound scenes may also be provided as un-labeled features. The sound scene classifier may run locally in each hearing instrument, it may be based on both hearing aids of a binaural hearing aid system, and/or be based on a sound scene classifier running on another device, such as a smartphone.

[0144] The hearing instrument (HD) further comprises a controller (CTR) configured to control the filter bank by applying different prototype filters to the analysis and synthesis filter banks in dependence of the current acoustic environment as classified by the sound scene classifier (SA), cf. output (PFW.sub.SSC) of the SA-CTR unit, e.g. feeding appropriate prototype filter coefficients to the respective analysis (FBA) and synthesis (FBS) filter banks. In other words, the hearing device is configured to have access to a database (e.g. stored in memory of the hearing device) comprising a multitude of different first and second prototype filters, e.g. filter coefficients thereof, together with a classification (e.g. a sound class) of the acoustic environment (or environments) where these are intended to be applicable, e.g.: [0145] h1(n), g1(n), Sound class1, [0146] h2(n), g2(n), Sound class2, [0147] . . . [0148] h.sub.Q(n), g.sub.Q(n), sound class Q.

[0149] where h.sub.q(n) and g.sub.q(n) represent first and second prototype filters of the analysis filter bank and the synthesis filter bank, respectively, for sound class q (acoustic environment #q). The number Q of sound classes having different prototype filters may e.g. be two or more, such as three or more, e.g. less than ten.

[0150] The prototype filters may depend on a hearing loss of the user. A prototype filter for a flat (across frequency) hearing loss may e.g. be configured to have less stopband attenuation and a narrower main lobe, whereas a prototype filter for a ski slope hearing loss (little loss at low frequency and high loss at high frequency) have more stop-band attenuation and a broader main lobe with more overlap between the neighboring bands.

[0151] It may thus make sense to individualize the prototype filters of the filter bank to a particular hearing loss even without changing the prototype filter across sound scenes (e.g. determined in a fitting session).

[0152] When a new (e.g. different from the previously detected) sound scene is detected, the controller (CTR) may be configured to (possibly instantly) fade from one prototype filter to another. Preferably the different adaptive prototype filters have the same group delay, such that the fading between the two filter banks maintain the same phase response (and the magnitude response changes are negligible). As only the prototype filter is changed, the frequency transformation in the filter banks (such as FFT or IFFT) can be re-used (cf. FIG. 3B).

[0153] The weights of a fixed beamformer may depend on the selected filter bank. Directional fixed beamformer weights may be changed depending on the selected prototype filter bank, see e.g. FIG. 7.

[0154] An adaptation rate of a feedback cancellation system may be temporarily increased when the filter banks are modified (i.e. when prototype filters are substituted).

[0155] A decision on changing the prototype filters may be applied to both hearing aids of a binaural hearing aid system simultaneously. The prototype filters may, however, be adapted on each hearing instrument separately.

[0156] FIG. 4B shows a second exemplary implementation of a hearing instrument with adaptive prototype analysis and synthesis filters. The embodiment of FIG. 4B is identical to the embodiment of FIG. 4A, apart from the scene detection unit (SA) receives the electric input signal (x) as (time-)frequency domain signals (X′). The embodiment of FIG. 4B comprises a specific analysis filter bank (FBA′) providing the (time domain) electric input signal (x) as a multitude of time variant frequency band signals (X′), which are fed to the scene detection unit (SA). The prototype filter of the specific analysis filter bank (FBA′) is not part of the adaptive adaptation to the acoustic environment (this is confined to the filter bank (FBA, FBS) of the forward audio path). The specific analysis filter bank (FBA′) may be configured to provide the same or a different number of frequency sub-bands than the filter bank (FBA, FBS) of the forward audio path.

[0157] An advantage of providing the input to the scene detection unit in the frequency domain is that it may be easier to extract scene dependent features from the frequency domain signal rather than from the time domain signal. Also, a higher frequency resolution may be used for the scene detection compared to the frequency resolution in the signal path used to generate the audio output signal, as the scene detection unit may utilize a longer latency than what is allowed in the signal path. The output (PFW.sub.SSC) of the SA-CTR unit is the same FIG. 4A to 4B. In both cases it provides/selects the appropriate prototype filter coefficients.

[0158] FIG. 5 shows an exemplary hearing aid according to the present disclosure. FIG. 5 shows an embodiment of a BTE-style hearing aid (HD) comprising an adaptive filter bank according to the present disclosure. The hearing device (HD) comprises a BTE-part and an ITE-part comprising an (possibly customized) ear mould or a more open dome-like structure (DO) or similar element e.g. for guiding the ITE-part in the ear canal of the user. The BTE-part (BTE) is adapted for being located at or behind an ear of a user, and the ITE-part (ITE) is adapted for being located in or at an ear canal of the user's ear. The ITE-part comprises a loudspeaker (HA-SPK) allowing sound to be played at the ear drum (Eardrum) of the user (cf. sound field S.sub.ED). The BTE-part and the ITE-part are electrically connected by connecting element (IC, e.g. an electric cable IC). The BTE-part comprises first and second input transducers, e.g. microphones (M.sub.BTE1 and M.sub.BTE2), respectively, which are used to pick up sound from the environment of a user wearing the hearing device (cf. sound field S). The ITE-part may comprise an environment facing microphone (M.sub.ENV), e.g. located at the entrance of the ear canal. The environment facing microphone has the advantage of picking up a signal that comprises the natural ‘Pinna cues’ reflecting the acoustic properties of Pinna. The ITE-part may further comprise an eardrum facing input transducer (MED, e.g. a microphone, or a vibration sensor) located so that it picks up sound or vibrations in or at the residual volume between the ITE-part and the ear drum (including from the speaker (HA-SPK) of the ITE-part) and provides an electric signal representative thereof. Such microphone may e.g. be used to various tasks for improving the processing of sound by the hearing aid, e.g. own voice detection or active noise cancellation (ANC). The connecting element (IC), e.g. an electric cable, is configured to comprise a multitude of electrically conducting wires or channels to allow the processor of the BTE part to communicate with the loudspeaker (HA-SPK), the environment facing microphone (M.sub.ENV) and/or the eardrum facing microphone (MED, if present), and possible other electronic components of the ITE part (ITE). Further, the electric cable may also be configured to allow energising the electronic components of the ITE-part (as well as those of the BTE-part) from the battery (BAT) of the BTE-part. The conductors of the electric cable are (e.g. via matching electric connectors on the cable and the BTE-part) connected to internal wiring in the BTE-part (cf. e.g. schematically illustrated as wiring Wx in the BTE-part) to relevant electronic circuitry of the hearing device, e.g. to the processor (DSP) and/or to a battery (BAT).

[0159] The hearing aid (HD) (here the BTE-part) further comprises two (e.g. individually selectable) wireless receivers (WLR.sub.1, WLR.sub.2) for providing respective directly received auxiliary audio input and/or control or information signals. The wireless receivers may be configured to receive signals from another hearing device (e.g. of a binaural hearing system) or from any other communication device, e.g. telephone, such as a smartphone, or from a wireless microphone or a T-coil. The wireless receivers may be capable of receiving (and possibly also of transmitting) audio and/or control or information signals. The wireless receivers may be based on Bluetooth or similar (short range communication) technology, e.g. UWB (Ultra Wide Band), or may be based on near-field communication (e.g. inductive coupling).

[0160] The hearing aid (HD) exemplified in FIG. 5 represents a portable device and further comprises a battery (BAT), e.g. a rechargeable battery, for energizing electronic components of the BTE-part and possibly the ITE-part.

[0161] The hearing aid (e.g. the processor (DSP)) may be 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 more frequency ranges to one or more other frequency ranges, e.g. to compensate for a hearing impairment of a user. The digital signal processor (DSP) may e.g. comprise appropriate filter banks (e.g. analysis as well as synthesis filter banks according to the present disclosure) to allow processing in the frequency domain (individual processing of frequency sub-band signals). The digital signal processor (DSP) may—as appropriate —comprise analogue to digital and digital to analogue converters (or a digital to digital) converters, for conversion of an analogue input signal and a processed (digital electric) signal to an analogue electric signal, respectively.

[0162] The BTE-part comprises a substrate SUB whereon a number of electronic components (MEM, FE, DSP) are mounted. The BTE-part comprises a configurable signal processor (DSP) and memory (MEM) accessible therefrom. The memory (MEM) may e.g. comprise filter coefficients for a multitude of different prototype filters of a filter bank according to the present disclosure, the prototype filters being configured for use in a corresponding multitude of different acoustic environments. In an embodiment, the signal processor (DSP) form part of an integrated circuit, e.g. a (mainly) digital integrated circuit. The BTE-part, e.g. the substrate, further comprises (mainly analogue) frontend-circuitry (FE) and radio-chips (WLR.sub.1, WLR.sub.2) as appropriate.

[0163] The partition of functional tasks between the BTE-part and the ITE-part may be different from the one mentioned in connection with the embodiments of FIG. 5. Some of the processing of the BTE-part and/or the ITE-part may be located in a dedicated processing device in communication with the BTE-part and/or the ITE-part, for example the processing related to the sound scene classification (cf. e.g. scene detection unit (SA) in FIG. 4A, 4B).

[0164] FIG. 6 shows a fourth exemplary hearing aid according to the present disclosure. The embodiment of FIG. 6 builds on the embodiment of FIG. 4A, comprising a (first) forward path comprising an input transducer (IT) (here a microphone) proving a time domain electric input signal (x) comprising audio, a (first) analysis filter bank (FBA1) for converting the time domain signal (x) to a (first) time-frequency signal (X.sub.1), a (first) processing unit (PRO1) for processing the (first) frequency sub-band signals (X.sub.1) to (first) processed frequency sub-band signals (Y.sub.1), a (first) synthesis filter bank (FBS1) for converting the (first) frequency sub-band signals (Y.sub.1) to a (first) processed time domain signal (y.sub.1), and an output transducer (OT) (here a loudspeaker) for converting a resulting time-domain output signal (y) to stimuli perceivable by the user as sound. The hearing device of FIG. 6 further comprises a second processing path connected between said input transducer (IT) and said output transducer (OT). The second processing path comprises a (second) analysis filter bank (FB2) providing a (second) time-frequency signal (X.sub.2), a (second) processing unit (PRO2) for processing the (second) frequency sub-band signals (X.sub.2) to (second) processed frequency sub-band signals (Y.sub.2), and a (second) synthesis filter bank (FBS2) for converting the (second) frequency sub-band signals (Y.sub.2) to a (second) processed time domain signal (y.sub.2). The outputs (y.sub.1, y.sub.2) of the respective first and second synthesis filter banks (FBS1, FBS2) are connected to a selector (SEL) for selecting one of the output streams (y.sub.1, y.sub.2) and presenting the selected (resulting) output stream (y) to the user via the output transducer (OT). The selector (SEL) is controlled in dependence of a currently determined (or selected) acoustic environment, cf. sound scene control signal (SSC), here provided by sound scene classifier (SA) in dependence of the electric into signal from the input transducer (IT). Instead of being located between the first and second synthesis filter banks (FBS1, FBS2) and the output transducer (OT), the selector (SEL) may be placed before (‘upstream of’) the synthesis filter banks, in which case we only need on synthesis filter bank. In the embodiment shown in FIG. 6, the two synthesis filter banks (FBS1, FBS2) may share the same synthesis filter bank coefficients (e.g. prototype filter coefficients).

[0165] An advantage of the proposed solution is that it allows different processing for at least two types of sound scenes, e.g. quiet and noisy. A fast switching of parameters from one scene to another may thereby by provided. Also, other parameters than the filter banks may be set in different ways depending on the at least two sound scenes. E.g. aggressiveness of the noise reduction system, beamformer weights, or smoothing time constants.

[0166] FIG. 7 shows a fourth exemplary hearing aid according to the present disclosure comprising beamformer. The embodiment of a hearing aid shown in FIG. 7 resembles the embodiment described in connection with FIG. 4A. The differences are described in the following. Instead of one input transducer shown in FIG. 4A, the embodiment of FIG. 7 comprises a multimode of N input transducers (IT.sub.1, . . . , IT.sub.N), e.g. microphones, each providing a (digitized electric input signal (x.sub.1, i=1, . . . , N) in the time domain from ‘Input sound’ (s1, . . . , s.sub.N) at the respective input transducers. Hence, the embodiment of FIG. 7 also comprises a multimode of N analysis filter banks (FBA), each providing respective frequency sub-band signals (X.sub.i, i=1, . . . , N). The hearing aid of FIG. 7 further comprises a beamformer (BF) connected to the multitude N of electric input signals (X.sub.1, . . . , X.sub.N) and configured to provide a beamformed signal Y.sub.BF in dependence of the N electric input signals and predefined (and/or adaptively updated) beamformer weights (BFW). The beamformer weights are adapted in dependence of the currently selected prototype filter of the filter bank (e.g. the analysis filter banks FBA)). The beamformed signal is fed to a signal processing unit (SP) for applying one or more processing algorithms to the beamformed signal (e.g. noise reduction (postfiltering), compressive amplification to compensate for a user's hearing impairment, etc.). The signal processing unit (SP) provides a processed signal Y in the time-frequency domain (as frequency sub-band signals). As in the embodiment of FIG. 4A, the processed signal Y is fed to an adaptive synthesis filter bank according to the present disclosure converting the processed signal Y to a corresponding time domain signal y, which is converted to stimuli perceivable as sound to the user by output transducer (OT), e.g. a loudspeaker or a vibrator providing ‘Output sound’.

[0167] The embodiment of FIG. 7 further comprises a sound scene classifier (SA) as described in connection with FIG. 4A, 4B, only her it receives N electric (time domain) input signals (x.sub.i, i=1, . . . , N)). Based thereon, the sound scene classifier (SA) provides a sound scene class control signal (SSC) indicative of a current acoustic environment. The hearing aid further comprises a controller for controlling the analysis filter banks (FBA) and the synthesis filter bank (FBS) by applying different first and second prototype filters to said at M analysis filter banks (FBA) and to the synthesis filter bank (FBS), respectively, in dependence of the sound scene class control signal (SSC) (indicative of a current acoustic environment). The hearing aid (HD) comprises a database of corresponding first and second prototype filters (h.sub.q(n), g.sub.q(n) and Sound Scene Class SSC.sub.q, q=1, . . . , Q), e.g. stored in memory (MEM) accessible to the controller (CTR). In response to a change in the currently determined Sound Scene Class (SSC), the controller (CTR) reads (cf. signal GetW.sub.SSC) the filter coefficients of the first (PF.sub.SW.sub.SSC) and second (PF.sub.SW.sub.SSC) prototype filters (for the analysis and synthesis filter banks, respectively) corresponding to the current Sound Scene Class. Thereby the filter coefficients of the first (PF.sub.SW.sub.SSC) and second (PF.sub.AW.sub.SSC) prototype filters are retrieved and forwarded to the respective analysis filter banks (FB) and the synthesis filter bank (FBS) (cf. bold arrows PF.sub.AW.sub.SSC to the M analysis filter banks (FBA) and PF.sub.SW.sub.SSC to the synthesis filter bank (FBS)). Further, different beamformer weights (BFWSSC) associated with at least some of the Sound Scene Class SSC.sub.q, q=1, . . . , Q, are also stored in memory (MEM). The weights of a fixed beamformer may depend on the selected prototype filter(s) of the filter bank. Fixed beamformer weights of the beamformer (BF) may be changed depending on the selected prototype filter of the filter bank.

[0168] The beamformer weights (BFW.sub.SSC) may be adapted in dependence of a current acoustic environment as described for the prototype filters of the adaptive filter bank: In response to a change in the currently determined Sound Scene Class (SSC), the controller (CTR) reads (cf. signal GetW.sub.SSC) the beamformer weights (BFW.sub.SSC) stored in memory corresponding to the current Sound Scene Class (SSC)/filter bank prototype filter. Thereby beamformer weights (BFW.sub.SSC) associated with the current prototype filters of the filter bank (and thus the current acoustic environment) are retrieved and forwarded to the beamformer (BF) for application instead of the previous beamformer weights (cf. bold arrow BFW.sub.SSC to the beamformer (BF)).

[0169] The currently determined Sound Scene Class (SSC) may be forwarded by the controller (CTR) to the signal processing unit (SSC), e.g. for use in one or more processing algorithms (e,g, noise reduction and/or level compression/gain estimation).

[0170] One or more, such as all of the sound scene classifier (SA), the controller (CTR), and the memory (MEM) may be located in another device than the hearing device, e.g. in an auxiliary device (cf. e.g. FIG. 8A, 8B), e.g. a dedicated processing device or a smartphone or remote control device. In such case the hearing aids and the auxiliary device must comprise appropriate transceiver circuitry to allow communication links with appropriate bandwidth and (low) latency to be stablished between the devices,

[0171] FIG. 8A shows a hearing system comprising a hearing aid and an auxiliary device in communication with each other.

[0172] FIG. 8B shows the auxiliary device of FIG. 8A configured to implement a user interface for the hearing aid by running an application program from which a mode of operation of the hearing aid can be selected.

[0173] FIGS. 8A and 8B together illustrate an exemplary application scenario of an embodiment of a hearing system (HD1, HD2, AD) according to the present disclosure.

[0174] FIG. 8A shows a hearing system comprising a hearing device (HD1, HD2), e.g. a hearing aid, and an auxiliary device (AD) in communication with each other. FIG. 8A shows an embodiment of a head-worn binaural hearing system comprising left and right hearing devices (HD1, HD2) in communication with each other and with a portable (handheld) auxiliary device (AD) functioning as a user interface (UI) for the binaural hearing aid system (see FIG. 8B). The binaural hearing system may comprise the auxiliary device AD (and the user interface UI). The binaural hearing system may comprise the left and right hearing devices (HD1, HD2) and be connectable to (but not include) the auxiliary device (AD). In the embodiment of FIG. 8A, the hearing devices (HD1, HD2) and the auxiliary device (AD) are configured to establish wireless links (WL-RF) between them, e.g. in the form of digital transmission links according to the Bluetooth standard (e.g. Bluetooth Low Energy, Ultra-Wideband (UWB), or equivalent technology). The links may alternatively be implemented in any other convenient wireless and/or wired manner, and according to any appropriate modulation type or transmission standard, possibly different for different audio sources.

[0175] The hearing devices (HD1, HD2) are shown in FIG. 8A as devices mounted at the ear (behind the ear) of a user (U). Other styles may be used, e.g. located completely in the ear (e.g. in the ear canal), fully or partly implanted in the head, etc. As indicated in FIG. 8A, each of the hearing devices may comprise a wireless transceiver to establish an interaural wireless link (IA-WL) between the hearing devices, e.g. based on inductive communication or RF communication (e.g. Bluetooth technology). Each of the hearing devices further comprises a transceiver for establishing a wireless link (WL-RF, e.g. based on radiated fields (RF)) to the auxiliary device (AD), at least for receiving and/or transmitting signals, e.g. control signals, e.g. information signals, e.g. including audio signals. The transceivers are indicated by RF-IA-Rx/Tx-2 and RF-IA-Rx/Tx-1 in the right (HD2) and left (HD1) hearing devices, respectively. The remote control-APP may be configured to interact with a single hearing device (instead of with a binaural hearing system, as illustrated in FIG. 8A).

[0176] The auxiliary device (AD) is adapted to run an application program, termed an APP, comprising executable instructions configured to be executed on the auxiliary device (e.g. a smartphone) to implement a user interface for the hearing device (or hearing system). The APP is configured to exchange data with the hearing device(s). FIG. 8B shows the auxiliary device (AD) of FIG. 8A configured to implement a user interface for the hearing device(s) (HD1, HD2) by running an application program from which a mode of operation of the hearing aid can be selected and via which selectable options for the user, and/or current status information can be displayed.

[0177] FIG. 8B illustrates the auxiliary device running an APP for configuring features of sound scene classification in the user's hearing aid or hearing aid system for use in an adaptive filter bank according to the present disclosure. An exemplary (configuration) screen of the user interface UI of the auxiliary device AD is shown in FIG. 8B. The user interface (UI) comprises a display (e.g. a touch sensitive display) displaying options for the user to manually set a sound scene class of the hearing aid or hearing aid system. The user interface (UI) is implemented as an APP on the auxiliary device (AD, e.g. a smartphone). The APP is denoted ‘Sound scene classifier’. Via the display of the user interface, the user (U) can accept an ‘Automatically detected sound scene’ (e.g. provided by sound scene classifier (SA), cf. e.g. FIG. 4A, 4B, 6, 7), cf. grey shaded box at the top of the screen. In the example, the automatically detected sound scene is ‘Speech in noise’. This can be accepted by pressing the solid black tick box (.square-solid.) in front of the automatically detected sound scene, and subsequently pressing the button ‘Activate classification’ at the bottom of the screen. Alternatively, the user is able to manually override the automatically detected sound scene, cf. the lower box denoted ‘Manual classification’, wherein a number of manually selectable sound scene classifications are listed (for which different prototype filters for the filter bank are available). The classification options for manual selection are: [0178] Car/bus/flight [0179] Broadband non-speech sounds/noise [0180] Party/competing voices [0181] Speech in noise [0182] Speech in silence [0183] Music [0184] Other

[0185] If the user wants to override the (default) automatically provided sound scene class, the tick box of the relevant class should be pressed (as indicated above and in the exemplary screen of the APP in FIG. 8B, Tarty/competing voices' has been selected as indicated by solid tick box (.square-solid.) and bold face letters). Subsequently the button ‘Activate classification’ at the bottom of the screen should be pressed.

[0186] The current acoustic environment may be automatically detected/classified by the hearing device(s) and/or by the auxiliary device (e.g. using acoustic features extracted from the electric input signals of the hearing device(s), and/or microphones and other sensors of the auxiliary device).

[0187] Further screens of the APP may allow the user to control other features of the hearing aid or hearing aid system, volume setting, program shift, monaural or binaural configuration, etc.

[0188] Switching between different screens of the APP may be achieved via left and right arrows in the bottom of the auxiliary device, or via ‘soft buttons’ integrated in the display of the user interface (UI).

[0189] In the embodiment of FIG. 8A, 8B, the auxiliary device (AD) is described as a smartphone. The auxiliary device may, however, be embodied in other portable electronic devices, e.g. an FM-transmitter, a dedicated remote control-device, a smartwatch, a tablet computer, etc.

[0190] In case of a binaural hearing aid system comprising first and second hearing aids in communication with each other, the two hearing aids may change prototype filters simultaneously based on a joint decision. Or the two hearing instruments may comprise different prototype filter coefficients. E.g. in asymmetric situations, where the noise level is much higher at on ear compared to the other ear. Also, if the hearing instrument user has asymmetric hearing loss, the listener may benefit between switching between different sets of prototype filters on one hearing instrument and another different set of prototype filter at the other hearing instrument.

[0191] 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.

[0192] 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 are not limited to the exact order stated herein, unless expressly stated otherwise.

[0193] 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.

[0194] 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.

REFERENCES

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