Abstract
A hearing device, e.g. a hearing aid, configured to be worn by a user, comprises a) two or more input transducers (e.g. microphones) wherein said two or more input transducers during use of the hearing device are arranged with a distance between them; b) a directional system comprising a directional algorithm configured to provide a directional pattern in dependence of said distance. The hearing device is configured to estimate a current value of said distance, or an equivalent acoustic delay, or beamformer weights of said directional system, thereby the directional performance can be optimized to the individual user.
Claims
1. A hearing device configured to be worn by a user, the hearing device comprising two or more input transducers, where said two or more input transducers, during use of the hearing device, are arranged with a distance between them, the two or more input transducers including one input microphone adapted for being located in the ear canal, and at least one input microphone adapted for being located behind the ear; and a directional system comprising a directional algorithm configured to provide a directional pattern in dependence of said distance, wherein the hearing device is configured to estimate a current value of said distance, or an equivalent acoustic delay, or beamformer weights of said directional system by measuring a phase difference of a sound signal originating from a sound outlet of the hearing device in the ear canal to the in-ear microphone and the behind the ear microphone.
2. A hearing device according to claim 1, wherein the directional algorithm is configured to emphasize sound from one direction and to suppress sound from other directions.
3. A hearing according to claim 1, wherein said two or more input transducers includes one microphone located in or at an earpiece and another located elsewhere on the body.
4. A hearing device according to claim 1, further comprising a loudspeaker.
5. A hearing device according to claim 1, wherein said distance is an acoustical microphone distance seen from an external sound field.
6. A hearing device according to claim 1, wherein said distance or an equivalent acoustic delay is used to optimize the directional algorithm, e.g. a delay and sum algorithm or an MVDR algorithm, for the individual user of the hearing device.
7. A hearing device according to claim 1, wherein the directional system provides a directional pattern designed to emphasize sound from one direction and to suppress sound from other directions.
8. A hearing device according to claim 7, wherein the directional pattern has a cancellation angle, or more cancellation angles, in the rear region that is dependent of the microphone distance.
9. A hearing device according to claim 1, further comprising a BTE-part adapted to be worn at or behind an ear of a user, and an ITE-part adapted to be located at or in an ear canal of the user, and wherein at least one input transducer is located in the BTE-part, wherein another input transducer is located in the ITE-part.
10. A hearing device according to claim 1, further comprising a time to time-frequency conversion unit allowing the processing of signals in the time-frequency domain.
11. A hearing device according to claim 1 configured to adaptively estimate said distance.
12. A method according to claim 11, wherein said distance may vary while wearing the hearing aid.
13. A hearing device according to claim 1 being constituted by or comprising a hearing aid, a headset, or an active ear protection device or a combination thereof.
14. A method according to claim 13, comprising adaptively estimating said distance.
15. A hearing device configured to be worn by a user, the hearing device comprising two or more input transducers or two microphones, where said two or more input transducers or two microphones, during use of the hearing device, are arranged with a distance between them; and a directional system comprising a directional algorithm configured to provide a directional pattern in dependence of said distance, wherein the hearing device is configured to estimate a current value of said distance, or an equivalent acoustic delay, or beamformer weights of said directional system and wherein the phase difference between the two or more input transducers or the two microphones of sound originating from the sound outlet is estimated by a loop gain estimation algorithm.
16. A hearing device according to claim 15, wherein the signal needed to estimate the loop gain comprises one or more pure tones or broadband noise.
17. A hearing device according to claim 15, wherein said estimate of the loop gain is provided in real time, in order to adaptively compensate for varying microphone distances during wear of the hearing device.
18. A method of operating a hearing device, the hearing device being configured to be worn by a user, the hearing device comprising two or more input transducers, where said two or more input transducers, during use of the hearing device, are arranged with a distance between them, and include one input microphone adapted to be located in the ear canal, and at least one input microphone adapted to be located behind the ear, the method comprising: estimating a current value of said distance, or an equivalent acoustic delay, or beamformer weights of a directional system by measuring a phase difference of a sound signal originating from a sound outlet of the hearing device in the ear canal to the in-ear microphone and the behind the ear microphone; and providing a directional pattern in dependence of said distance.
19. A method of operating a hearing device, the hearing device being configured to be worn by a user, the hearing device comprising two or more input transducers, where said two or more input transducers, during use of the hearing device, are arranged with a distance between them, the method comprising: estimating a current value of said distance, or an equivalent acoustic delay, or beamformer weights of a directional system; estimating a phase difference between the two or more input transducer or two microphones of sound originating from the sound outlet using a loop gain estimation algorithm; and providing a directional pattern in dependence of said distance.
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 schematically shows basic elements of a first embodiment of a hearing device comprising a near-field beamformer implementing a feedback suppression system according to the present disclosure;
(3) FIG. 1B schematically shows basic elements of a second embodiment of a hearing device comprising a near-field beamformer implementing a feedback suppression system according to the present disclosure;
(4) FIG. 1C schematically shows basic elements of a third embodiment of a hearing device comprising a near-field beamformer implementing a feedback suppression system according to the present disclosure; and
(5) FIG. 1D schematically shows basic elements of a fourth embodiment of a hearing device comprising a near-field beamformer implementing a feedback suppression system according to the present disclosure;
(6) FIG. 2A schematically shows basic elements of a first embodiment of a hearing device comprising a feedback suppression system and a far-field beamformer filtering unit according to the present disclosure; and
(7) FIG. 2B schematically shows basic elements of a second embodiment of a hearing device comprising a feedback suppression system and a far-field beamformer filtering unit according to the present disclosure,
(8) FIG. 3 shows an embodiment of a RITE-type hearing device according to the present disclosure comprising a BTE-part, an ITE-part and a connecting element,
(9) FIG. 4A shows an embodiment of a hearing device according to the present disclosure comprising a BTE-part located behind an ear (as seen from above) and comprising a microphone and an ITE-part located in the ear canals comprising microphone and a loudspeaker, and
(10) FIG. 4B illustrates a scenario comprising the hearing device of FIG. 4A located in the acoustic far-field of a relatively distant sound source and in the acoustic near-field of a relatively close sound source,
(11) FIG. 5 shows an embodiment of a (far-field) beamformer filtering unit for use in a hearing device according to the present disclosure,
(12) FIG. 6A shows a first embodiment of a hearing device comprising a far-field beamformer according to the present disclosure, and
(13) FIG. 6B shows a second embodiment of a hearing device comprising a far-field beamformer according to the present disclosure, and
(14) FIG. 7A schematically shows a difference in magnitude vs. frequency of a sound signal originating from the output transducer and arriving at the ITE and BTE-microphones, respectively, and
(15) FIG. 7B schematically shows a difference in phase vs. frequency of a sound signal originating from the output transducer and arriving at the ITE and BTE-microphones, respectively.
(16) 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.
(17) 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
(18) 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.
(19) 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.
(20) It is a general known problem for hearing aid users that acoustical feedback from the ear canal causes the hearing aid to whistle if the gain is too high and/or if the vent opening in the ear mould is too large. The more gain that is needed to compensate for the hearing loss, the smaller the vent (or effective vent area) must be to avoid whistle, and for severe hearing losses even the leakage between the ear mould (without any deliberate vent) and the ear canal can cause the whistling.
(21) Hearing aids with microphones behind the ear can achieve the highest gain, due to their relatively large distance from the ear canal and vent in the mould. But for users with severe hearing loss needing high gain, it can be difficult to achieve a sufficient venting in the mould (with an acceptable howl risk).
(22) EP2849462A1 proposes to solve the conflicting demands of good sound quality and good directionality by combining one or more supplementary microphones, e.g. located in a shell or housing of a BTE (Behind-The-Ear) hearing assistance device while introducing an audio microphone in pinna, e.g. at the entrance to the ear canal. The audio microphone is preferably the main input transducer and the signal coming from it treated according to control signals originating from the supplementary microphone(s).
(23) EP2843971A1 deals with a hearing aid device comprising an “open fitting” providing ventilation, a receiver arranged in the ear canal, a directional microphone system comprising two microphones arranged in the ear canal at the same side of the receiver, and means for counteracting acoustic feedback on the basis of sound signals detected by the two microphones. An improved feedback reduction can thereby be achieved, while allowing a relatively large gain to be applied to the incoming signal.
(24) FIG. 1A-1D shows four embodiments of a hearing device (HD), e.g. a hearing aid, according to the present disclosure. Each of the embodiments of a hearing device (HD) comprises a forward path between an input unit (IU; IUa, IUb) for providing a multitude of electric input signals representing sound, and an output unit (OU) for converting a processed signal to a stimulus perceivable by the user as sound. The hearing device further comprises a feedback suppression unit (FBC) for suppressing (e.g. cancelling) feedback from the output unit to the input unit and providing a feedback corrected signal IN.sub.FBC. Each of the four embodiments of a hearing device (HD) further (optionally) comprises a signal processor (HLC) for applying one or more signal processing algorithms to a signal of the forward path (e.g. a compressive amplification algorithm for compensating for a user's hearing impairment). The feedback suppression system (FBC) may e.g. be implemented as a near-field beamformer, as indicated in FIG. 1A by reference ‘Near-field beamfomer’ at the feedback suppression system (FBC).
(25) In the embodiment of FIG. 1A, the input unit (IUa, IUb) comprises a first input transducer (IT1, e.g. a microphone) for picking up a sound signal from the environment and providing a first electric input signal (IN1), and a second input transducer (IT2) for picking up a sound signal from the environment and providing a second electric input signal (IN2). The second input transducer (IT2) is adapted for being located in an ear of a user, e.g. near the entrance of an ear canal (e.g. at or in the ear canal or outside the ear canal but in the concha part of pinna). The aim of the location is to allow the second input transducer to pick up sound signals that include the cues resulting from the function of pinna (e.g. directional cues) and to allow an estimate of feedback to be provided.
(26) The embodiment of FIG. 1A comprises two input transducers (IT1, IT2). The number of input transducers may be larger than two ((IT1, . . . , ITn), n being any size that makes sense from a signal processing point of view), and may include input transducers of a mobile device, e.g. a smartphone or even fixedly installed input transducers in communication with the hearing device.
(27) The embodiments of FIGS. 1B, 1C and 1D comprise the same functional units as the embodiment of FIG. 1A (units IU (IT1, IT2), FBC, HLC, and OU). In the embodiments of FIGS. 1B, 1C and 1D, the input unit (IU) comprises first and second input transducers in the form of first and second microphones M.sub.BTE and M.sub.ITE, e.g. located behind an ear and at or in an ear canal, respectively, providing first and second electric input signals IN.sub.BTE and IN.sub.ITE, respectively, and the output unit (OU) comprises an output transducer in the form of a loudspeaker (SPK) for converting a processed electric output signal OUT from the processor (HLC) to an acoustic signal (e.g. vibrations in air). Alternatively, the output transducer may comprise a vibrator for delivering stimuli to bone of the head of the user (to implement a bone conducting hearing device). In the embodiments of FIGS. 1B, 1C and 1D, different embodiments of the feedback suppression unit (FBC) are schematically illustrated.
(28) The embodiments of FIGS. 1B, 1C and 1D comprise different embodiments of the feedback suppression unit (FBC).
(29) FIG. 1B shows an embodiment of a hearing device (HD) as shown in FIG. 1A, but where the feedback suppression unit (FBC)—indicated in the dashed enclosure—comprises a feedback estimation unit (FBE) for estimating feedback from the output unit (OU), here loudspeaker (SPK) to the input unit (here microphone M.sub.BTE). The feedback estimation unit (FBE) comprises adjustment unit (ADU) for modifying the second electric input signal IN.sub.ITE in correspondence with an acoustic transfer function, or an impulse response, from the second input transducer (microphone M.sub.ITE) to the first input transducer (microphone M.sub.BTE) and providing a modified second electric input signal FB.sub.est representative of an estimate of the feedback. The feedback suppression unit (FBC) further comprises a combination unit (here sum unit ‘+’) for combining the second electric input signal FB.sub.est with the first electric input signal IN.sub.BTE and providing a feedback corrected input signal IN.sub.FBC that is fed to the processor (HLC). In the embodiment of FIG. 1B, the second electric input signal representative of an estimated feedback FB.sub.est is subtracted from the first electric input signal IN.sub.BTE resulting in the feedback corrected input signal IN.sub.FBC. The adjustment unit (ADU) may be implemented by predetermined (e.g. frequency dependent) acoustic transfer functions (or impulse responses) or adaptively determined acoustic transfer functions (or impulse responses), as e.g. indicated in FIG. 1D. The adjustment unit (ADU) may be implemented by (predetermined or adaptively determined) complex weights representing appropriate (e.g. frequency dependent) phase changes (delays) and attenuation. In an embodiment, the adaptively determined acoustic transfer functions (or impulse responses) are determined in connection with a start-up of the hearing device (typically at least once a day for a hearing aid).
(30) FIG. 1C shows an embodiment of a hearing device (HD) as shown in FIG. 1B, but where the feedback estimation unit (FBE) additionally receives the first electric input signal IN.sub.BTE and the processed electric output signal OUT as inputs. Thereby an adaptive estimation of the feedback can be implemented (by adaptively estimating a transfer function from the second to the first input transducer). An example of this is illustrated in FIG. 1D.
(31) In FIG. 1D shows an embodiment of a hearing device (HD) as shown in FIG. 1C, but where the feedback estimation unit (FBE) is further exemplified. The feedback estimation unit (FBE) (enclosed by dotted outline in FIG. 1D) providing an estimate FB.sub.est of the feedback from the loudspeaker (SPK) to the BTE-microphone (M.sub.BTE) comprises adjustment unit (ADU) and control unit (CTR). The adjustment unit (ADJ) comprises delay unit (D) for applying a delay to the second electric input signal IN.sub.ITE corresponding to the delay of the acoustic propagation path of sound from the ITE to the BTE microphone, and gain unit (G) for applying an attenuation to the second electric input signal IN.sub.ITE corresponding to the attenuation of the acoustic propagation path of sound from the ITE to the BTE microphone. The control unit (CTR) is configured to adaptively control the delay and gain estimation units in dependence of the respective electric input signals IN.sub.BTE and IN.sub.ITE and the output signal (OUT) to the loudspeaker (SPK). In an embodiment, the control unit (CTR) is configured to estimate the difference in delay between the reception of a given signal from the loudspeaker at the two microphones (M.sub.BTE and M.sub.ITE). A variety of methods may be applied, e.g. performing a pure tone sweep (e.g. by a generator of the processor (HLC)), where the phase difference in the signal picked up by the microphones are determined (e.g. in the control unit (CTR). The thus estimated current delay difference (D.sub.BTE−D.sub.ITE) can be applied to the second electric signal IN.sub.ITE by the delay unit (D) (controlled by the control unit (CTR)). Alternatively, the processor can be configured to issue a ping type signal, and the time difference between the arrival of the ‘ping’ at the two microphones (M.sub.BTE and M.sub.ITE) can be determined by the control unit (CTR). In an embodiment, the control unit (CTR) comprises respective level detection units for estimating a current level (L.sub.BTE and L.sub.ITE) of the first and second electric input signals (IN.sub.BTE and M.sub.ITE). A current level difference (L.sub.ITE−L.sub.BTE) can thus be determined and a corresponding attenuation applied to the second electric signal IN.sub.ITE by the gain estimation unit (G) (controlled by the control unit (CTR).
(32) The second input transducer (IT2; M.sub.ITE in FIG. 1A-1D) and the output unit (OU), e.g. output transducer (OT, SPK) are e.g. located in an in-the-ear part (ITE) adapted for being located in the ear of a user, e.g. at or in the ear canal of the user, e.g. as is customary in a RITE-type hearing device. Alternatively, the second input transducer (IT2; M.sub.ITE) may be located in concha, e.g. in the cymba-region. The processor (HLC) may be located in a separate body-worn part, e.g. in a so-called BTE-part adapted for being located at or (at least partially) behind pinna. Alternatively, the processor (HLC) may be located elsewhere, e.g. in the ITE-part (ITE) or in another part in communication with the input and output units, e.g. in a separate processing part, e.g. a smartphone or similar device. The first input transducer (IT1; M.sub.BTE) may e.g. be located in the behind-the-ear part (BTE) or elsewhere on the head of the user, e.g. at an ear of the user.
(33) The ‘operational connections’ between the functional elements of the hearing device (HD) (units IU (IT1, IT2), FBC, HLC, and OU) can be implemented in any appropriate way allowing signals to the transferred (possibly exchanged) between the elements (at least to enable a forward path from the input unit (transducers) to the output unit (transducer), via (and possibly in control of) the processor (HLC)). The different units of the hearing device may be electrically connected via wired electric connections. Alternatively, non-wired electric connections, e.g. wireless connections, e.g. based on electromagnetic signals, may be used. In such case the inclusion of relevant antenna and transceiver circuitry is implied. One or more of the wireless links may be based on Bluetooth technology (e.g. Bluetooth Low-Energy or similar technology). Thereby a relatively large bandwidth and a relatively large transmission range is provided. Alternatively or additionally, one or more of the wireless links may be based on near-field, e.g. capacitive or inductive, communication. The latter has the advantage of having a low power consumption.
(34) The processor (HLC) is configured to process the feedback corrected signal IN.sub.FBC (or a processed version thereof), and for providing a processed (preferably enhanced) output signal (OUT). The processor (HLC) may comprise a number of processing algorithms, e.g. a noise reduction algorithm, for enhancing the feedback corrected (e.g. beamformed and optionally further noise reduced) signal, e.g. according to a user's needs (e.g. to compensate for a hearing impairment) to provide the processed output signal (OUT). All embodiments of a hearing device are adapted for being arranged at least partly on a user's head or at least partly implanted in a user's head (an at least partly implanted part e.g. comprising a carrier for attaching a vibrator of a bone-conduction hearing device).
(35) The embodiments of a hearing device (HD) of FIGS. 2A and 2B comprises the same functional elements as described in FIG. 1A-1D. A difference is that the embodiments of FIGS. 2A and 2B, each comprises three input transducers (M.sub.BTE1, M.sub.BTE2, M.sub.ITE) in the form of microphones (e.g. omni-directional microphones). Each of the input transducers of the input unit can theoretically be of any kind, such as comprising a microphone (e.g. a normal microphone or a vibration sensing bone conduction microphone), or an accelerometer, or a wireless receiver. Each of the embodiments of a hearing device (HD) comprises an output unit (OU) comprising an output transducer (OT) for converting a processed output signal to a stimulus perceivable by the user as sound. In the embodiments of a hearing device (HD) of FIGS. 1B, 1C, 1D, and 2A and 2B, the output transducer is shown as receivers (loudspeakers, SPK). A receiver can e.g. be located in an ear canal (RITE-type (Receiver-In-The-ear) or a CIC (completely in the ear canal-type) hearing device) or outside the ear canal (e.g. in a BTE-type hearing device), e.g. coupled to a sound propagating element (e.g. a tube) for guiding the output sound from the receiver to the ear canal of the user (e.g. via an ear mould located at or in the ear canal). Alternatively, other output transducers can be envisioned, e.g. a vibrator of a bone anchored hearing device.
(36) The embodiments of a hearing device (HD) of FIG. 1A-1D, and FIG. 2A-2B are shown without indication of any domain transformations of the electric input and processed signals. In general, at least a transformation from analogue to digital domain is implied (e.g. using appropriate analogue to digital converters e.g. forming part if the respective input transducers (e.g. microphones) or included as separate units. The signal processing may be performed fully or partially in the time domain. In an embodiment, the hearing device comprises appropriate time to frequency conversion units (t/f) enabling analysis and/or processing of the electric input signals (IN.sub.BTE1, IN.sub.BTE2, IN.sub.ITE) from the input transducers (here microphones M.sub.BTE1, M.sub.BTE2, M.sub.ITE), respectively, in the frequency domain. In the embodiments of FIGS. 2A and 2B, the time-frequency conversion units may be included in the beamforming filtering unit (BF, for signals IN.sub.BTE1, IN.sub.BTE2, and possibly IN.sub.ITE) and in the feedback suppression system (FBC, for signal IN.sub.ITE), but may alternatively form part of the respective input transducers or of the signal processor (HLC) or be separate units. The hearing device (HD) may further comprise a frequency to time conversion unit (f/t), e.g. included in the signal processor (HLC) or be located elsewhere, e.g. in connection with the output unit, e.g. the output transducer (OT).
(37) FIG. 2A shows an embodiment of a hearing device (HD) as shown in FIG. 1C. In addition, the embodiment of FIG. 2A comprises a beamformer filtering unit (BF, denoted Far-field beamformer) for providing a spatially filtered (beamformed) signal IN.sub.BF, which is fed to the feedback suppression unit (FBC, denoted Near-field beamformer) and processed as described in FIG. 1C. The (far-field) beamformer filtering unit (BFU) is e.g. configured to maintain (or attenuate less) signal components in the sound field around the (first) microphones (M.sub.BTE1, M.sub.BTE2) from a direction to a current target sound source (e.g. S.sub.FF in FIG. 4B), while signal components from other directions are attenuated (e.g. attenuated more than signals from the target direction). The (far-field) beamformer filtering unit (BFU) may e.g. comprise a beamformer as described in FIG. 5.
(38) FIG. 2B shows an embodiment of a hearing device (HD) as shown in FIG. 2A. In addition, the embodiment of FIG. 2B the feedback estimation unit (FBE) further receives the (first) electric input signals (IN.sub.BTE1, IN.sub.BTE2) from the first and second (BTE) microphones (M.sub.BTE1, M.sub.BTE2). The feedback estimate (FB.sub.est) is thus dependent of all three electric input signals ((IN.sub.BTE1, IN.sub.BTE2, IN.sub.ITE), the beamformed signal (IN.sub.BF) and the processed electric output signal (OUT). The resulting feedback estimate (FB.sub.est) that is fed to the combination unit (‘+’) is e.g. high pass filtered (cf. indication ‘HP’ on the output from the feedback estimation unit (FBE)). The high pass filtering of the ITE microphone signal (IN.sub.ITE) is intended to focus on the frequencies, where feedback is known to occur (i.e. above 1 kHz, e.g. in a range between 1 kHz and 8 kHz, such as between 1 kHz and 4 kHz). Further, the beamformer filtering unit (BFU) receives (a possibly low pass filtered version of (cf. indication ‘LP’ on the input to the beamformer filtering unit (BF))) the (second) electric input signal (IN.sub.ITE), so that the beamformed signal IN.sub.BF is based on a combination of the three input signals (IN.sub.BTE1, IN.sub.BTE2, and (e.g. low pass filtered) IN.sub.ITE)). The low pass filtering of the ITE microphone signal (IN.sub.ITE) is intended to focus on the frequencies, where feedback is known NOT to occur.
(39) The directional system (beamformer filtering unit BFU) may e.g. comprise a low frequency part and a high frequency part. At relatively low frequencies, e.g. below 1 kHz or below 1.5 kHz, the beamformer filtering unit relies on a combination of a signal from the ITE-microphone (IN.sub.ITE) and one or both of the signals from the BTE microphones (IN.sub.BTE1, IN.sub.BTE2). At relatively high frequencies, e.g. above 1 kHz or above 1.5 kHz, the beamformer filtering unit relies only on the signals from the BTE microphones (IN.sub.BTE1, IN.sub.BTE2).
(40) FIG. 3 shows an embodiment of a hearing device according to the present disclosure. The hearing device (HD), e.g. a hearing aid, is of a particular style (sometimes termed receiver-in-the ear, or RITE, style) comprising a BTE-part (BTE) adapted for being located at or behind an ear of a user, and an ITE-part (ITE) adapted for being located in or at an ear canal of the user's ear and comprising a receiver (loudspeaker).
(41) The BTE-part and the ITE-part are connected (e.g. electrically connected) by a connecting element (IC) and internal wiring in the ITE- and BTE-parts (cf. e.g. wiring Wx in the BTE-part).
(42) In the embodiment of a hearing device in FIG. 3, the BTE part comprises an input unit (IU in FIG. 1A-1C) comprising two (first) input transducers (e.g. microphones) (M.sub.BTE1, M.sub.BTE2), each for providing an electric input audio signal representative of an input sound signal (S.sub.BTE) (originating from a sound field S around the hearing device). The input unit further comprises two wireless receivers (WLR.sub.1, WLR.sub.2) for providing respective directly received auxiliary audio and/or control input signals (and/or allowing transmission of audio and/or control signals to other devices). The hearing device (HD) comprises a substrate (SUB) whereon a number of electronic components are mounted, including a memory (MEM) e.g. storing different hearing aid programs (e.g. parameter settings defining such programs) and/or hearing aid configurations, e.g. input source combinations (M.sub.BTE1, M.sub.BTE2, W.sub.LR1, W.sub.LR2), e.g. optimized for a number of different listening situations. The substrate further comprises a configurable signal processor (DSP, e.g. a digital signal processor, including the processor (HLC), feedback suppression (FBC) and beamformers (BFU) and other digital functionality of a hearing device according to the present disclosure). The configurable signal processing unit (DSP) is adapted to access the memory (MEM) and for selecting and processing one or more of the electric input audio signals and/or one or more of the directly received auxiliary audio input signals, based on a currently selected (activated) hearing aid program/parameter setting (e.g. either automatically selected, e.g. based on one or more sensors and/or on inputs from a user interface). The mentioned functional units (as well as other components) may be partitioned in circuits and components according to the application in question (e.g. with a view to size, power consumption, analogue vs. digital processing, etc.), e.g. integrated in one or more integrated circuits, or as a combination of one or more integrated circuits and one or more separate electronic components (e.g. inductor, capacitor, etc.). The configurable signal processor (DSP) provides a processed audio signal, which is intended to be presented to a user. The substrate further comprises a front end IC (FE) for interfacing the configurable signal processor (DSP) to the input and output transducers, etc., and typically comprising interfaces between analogue and digital signals. The input and output transducers may be individual separate components, or integrated (e.g. MEMS-based) with other electronic circuitry.
(43) The hearing device (HD) further comprises an output unit (e.g. an output transducer) providing stimuli perceivable by the user as sound based on a processed audio signal from the processor (HLC) or a signal derived therefrom. In the embodiment of a hearing device in FIG. 3, the ITE part comprises the output unit in the form of a loudspeaker (receiver) for converting an electric signal to an acoustic (air borne) signal, which (when the hearing device is mounted at an ear of the user) is directed towards the ear drum (Ear drum), where sound signal (S.sub.ED) is provided. The ITE-part further comprises a guiding element, e.g. a dome, (DO) for guiding and positioning the ITE-part in the ear canal (Ear canal) of the user. The ITE-part further comprises an (second) input transducer, e.g. a microphone (M.sub.ITE), for providing an electric input audio signal UNITE in FIG. 1A-D, 2A-B) representative of an input sound signal (S.sub.ITE).
(44) The hearing device (HD) exemplified in FIG. 3 is a portable device and further comprises a battery (BAT), e.g. a rechargeable battery, e.g. based on Li-Ion battery technology, e.g. for energizing electronic components of the BTE- and possibly ITE-parts. In an embodiment, the hearing device, e.g. a hearing aid (e.g. the processor (HLC)), is adapted to provide a frequency dependent gain and/or a level dependent compression and/or a 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.
(45) FIG. 4A shows an embodiment of a hearing aid (HD) according to the present disclosure comprising a BTE-part (BTE) located behind an ear (Pinna, as seen from above) and comprising a microphone (M.sub.BTE) and an ITE-part (ITE) located in the ear canal (Ear canal) comprising a microphone (M.sub.ITE) and a loudspeaker (SPK). The microphone (M.sub.ITE) faces the environment. The loudspeaker (SPK) faces the ear drum (cf. Ear drum in FIG. 4B).
(46) The dashed lines in FIG. 4A indicate the propagation of the external sound field approaching from the frontal direction (Far-field sound)(- - - -) and the sound field generated by the speaker in the ear canal (Near-field sound)(- - - - -) The path length difference for sound arriving at the microphones of the hearing device originating from the far field and from the near-field, respectively, may be substantial.
(47) The (far-field) directional microphone system is designed to emphasize sound from one direction (typically frontal) and suppress sound from other directions, (usually sounds from behind). The directional pattern typically has a cancellation angle (or more cancellation angles) in the rear region (e.g. adaptively determined) that is dependent of the microphone distance. In a simple way this may be achieved by delaying the signal from one microphone and then subtracting the two microphone signals. The delay depends on the microphone distance and the desired direction of the cancellation angle. The microphone distance needed by the algorithm is the acoustical microphone distance seen from the external sound field. Alternatively, the far-field directional system (beamformer filtering unit) may comprise a linearly constrained minimum variance (LCMV) beamformer, e.g. a minimum variance distortionless response (MVDR) beamformer.
(48) In an embodiment, the hearing device, e.g. a hearing instrument, estimates the microphone distance by measuring the phase difference of a sound signal originating from the sound outlet of the hearing device (e.g. loudspeaker SPK in FIG. 4A) in the ear canal to the in-ear microphone (M.sub.ITE) and the behind the ear microphone (M.sub.BTE). This can be used to calculate the acoustical microphone distance from sound originating from the ear. This distance correlates to the microphone distance for external sound fields (cf. FIG. 4A), and can then be used to optimize the directional algorithm for the individual user.
(49) The algorithm used to estimate the phase difference between the two microphone of sound originating from the sound outlet, can be a loop gain estimation algorithm, usually used to estimate the feedback path for minimizing the undesired acoustical feedback. The signal needed to estimate the loop gain may e.g. either be pure tones or broadband noise. This kind of system may also estimate the loop gain real time, in order to adaptively compensate for varying microphone distances during wear.
(50) Alternatively, the signal to estimate the delay difference between the two microphones can be broadband noise, pure tone sweep where the phase difference in the signal picked up by the microphones are determined. Alternatively, the signal could be of a ping type where the time delay is measured by the two microphones.
(51) FIG. 4B schematically illustrates a scenario comprising the hearing device (HD) of FIG. 4A located in the acoustic far-field (denoted S.sub.BTE-FF and S.sub.ITE-FF at the BTE and ITE microphones, M.sub.BTE1, M.sub.BTE2 and M.sub.ITE, respectively) of a relatively distant sound source (S.sub.FF) and in the acoustic near-field (denoted S.sub.BTE-NF and S.sub.ITE-NF at the BTE and ITE microphones, respectively) of a relatively close sound source (S.sub.NF). ‘Relatively close’ and ‘relatively distant’ is taken relative to the hearing device (microphones). In the scenario of FIG. 4B, the relatively close sound source (S.sub.NF) originates from sound played by the loudspeaker (SPK) located in the ear canal (Ear canal) of the user. The sound S.sub.ED is reflected by the walls and ear drum (Ear drum) of the ear canal and propagated towards the environment arriving at the ITE-microphone (M.sub.ITE) and later (farther away) at the first and second BTE-microphones (M.sub.BTE1, M.sub.BTE2). The acoustic far-field (S.sub.BTE-FF and S.sub.ITE-FF at the BTE and ITE microphones, respectively) is illustrated by straight solid lines illustrating the plane wave nature of sound waves in the far-field approximation. The acoustic near-field (S.sub.BTE-NF and S.sub.ITE-NF at the BTE and ITE microphones, respectively) is illustrated by curved dashed lines illustrating the non-parallel wave fronts of sound waves in the near-field approximation. In the near-field, acoustic intensity can vary greatly with distance, whereas in the far-filed, it has a (smaller) constant decrease (in a logarithmic representation, 6 dB each time the distance from the source is doubled). The S.sub.ITE-FF part of the signal picked up by M.sub.ITE is nearly the same as the S.sub.BTE-FF part of the signal from the far-field sound source, but the attenuation G.sub.ITE-BTE applied to the total signal picked up by the ITE-microphone by the adjustment unit (cf. e.g. FIG. 1D) is relatively large, so the (attenuated) component is insignificant compared to the S.sub.BTE-FF part received at the BTE-microphone(s) (i.e. IN.sub.BTE-FF>>G.sub.ITE-BTE*IN.sub.ITE-FF, where IN.sub.ITE=IN.sub.ITE-FF+IN.sub.ITE-NF, and IN.sub.IBTE=IN.sub.BTE-FF+IN.sub.BTE-NF). Since IN.sub.BTE-NF=FB and FB.sub.est=G.sub.ITE-BTE*IN.sub.ITE=G.sub.ITE-BTE*(IN.sub.ITE-FF+IN.sub.ITE-NF), and IN.sub.BTE-FF is approximated by IN.sub.BTE-FB.sub.est, IN.sub.BTE-FF˜IN.sub.BTE G.sub.ITE-BTE*(IN.sub.ITE-FF+IN.sub.ITE-NF). To minimize such error (improve the feedback estimate), the term CITE-BTE*IN.sub.ITE-FF may be adaptively estimated and compensated for (cf. e.g. FIG. 6A, 6B).
(52) The feedback path transfer functions which represent the change of the acoustical sound signal from the speaker SPK to each of the microphones (M.sub.ITE and M.sub.BTEx, x=1, 2) are e.g. denoted H.sub.ITE and H.sub.BTEx, respectively. The relative feedback path transfer function between the ITE and BTE microphones (M.sub.ITE and M.sub.BTEx, x=1, 2) is given by the ratio between H.sub.BTEx and H.sub.ITE. Similarly, the transfer functions from far-field sound source S.sub.FF to each of the microphones (M.sub.ITE and M.sub.BTEx, x=1, 2) are denoted A.sub.BTEx and A.sub.ITE, respectively. When the sound source S.sub.FF is far from the user (microphones), it is expected that the ratio between the transfer functions A.sub.BTEx and A.sub.ITE is smaller than the ratio between the feedback path transfer functions H.sub.BTEx and H.sub.ITE, respectively, because the feedback path transfer functions are present in the acoustic near field, where the relative difference in the distance between the microphones M.sub.ITE and M.sub.BTEx to the speaker SPK (S.sub.NF) is greater than the relative difference in the distance between the microphones M.sub.ITE and M.sub.BTEx to the far-field sound source S.sub.FF, i.e., (|A.sub.ITE|/|A.sub.BTEx|)<(|H.sub.ITE|/|H.sub.BTEx|), as further discussed in EP2947898A1 (cf. section [0076] regarding FIG. 4).
(53) The distance between the near field sound source S.sub.NF (the loudspeaker SPK) and the ITE-microphone M.sub.ITE may e.g. be of the order of 0.02 m. The distance between the near field sound source S.sub.NF (the loudspeaker SPK) and each of the BTE-microphones (M.sub.BTEx, x=1, 2) may e.g. be of the order of 0.07 m. The difference in distance between the ITE and BTE microphones may e.g. be of the order of 0.05 m. The distance between the far-field sound source S.sub.FF (e.g. a communication partner) and the user (i.e. any of the microphones (M.sub.ITE and M.sub.BTEx, x=1, 2)) may e.g. be of the order of 1 m or more.
(54) FIG. 5 shows an embodiment of a (far-field) beamformer filtering unit for use in a hearing device according to the present disclosure. An exemplary beamformer filtering unit (BFU) as indicated in FIGS. 2A and 2B is outlined in the following with reference to FIG. 5. FIG. 5 shows a part of a hearing aid comprising first and second microphones (M.sub.BTE1 M.sub.BTE2) providing respective first and second electric input signals IN.sub.BTE1 and IN.sub.BTE2, respectively and a beamformer filtering unit (BFU) providing a beamformed signal IN.sub.BF based on the first and second electric input signals. A direction from the target signal to the hearing aid is e.g. defined by the microphone axis and indicated in FIG. 5 by arrow denoted Target sound. The target direction can be any direction, e.g. a direction to the user's mouth (to pick up the user's own voice), or a direction to a communication partner in front of the user. An adaptive beam pattern (Y (Y(k))), for a given frequency band k, k being a frequency band index, is obtained by linearly combining an omnidirectional delay-and-sum-beamformer (O (O(k))) and a delay-and-subtract-beamformer (C (C(k))) in that frequency band. The adaptive beam pattern arises by scaling the delay-and-subtract-beamformer (C(k)) by a complex-valued, frequency-dependent, adaptive scaling factor β(k) (generated by beamformer ABF) before subtracting it from the delay-and-sum-beamformer (O(k)), i.e. providing the beam pattern Y,
Y(k)=O(k)−β(k)C(k).
(55) It should be noted that the sign in front of β(k) might as well be +, if the sign(s) of the weights constituting the delay-and-subtract beamformer C is/are appropriately adapted. Further, β(k) may be substituted by β*(k), where * denotes complex conjugate, such that the beamformed signal IN.sub.BF is expressed as IN.sub.BF=(w.sub.o(k)−β(k).Math.w.sub.c(k)).sup.H.Math.IN(k), where IN(k)=(IN.sub.BTE1(k), IN.sub.BTE2(k)).
(56) A beamformer filtering unit of this nature is e.g. further described in EP2701145A1, and in EP3236672A1. Other kinds of beamformer filtering units may be used, though.
(57) FIG. 6A shows a first embodiment of a hearing device (HD) comprising a far-field beamformer unit (BF) according to the second aspect of the present disclosure. The hearing device comprises a BTE-part and an ITE part adapted for being located at or behind pinna and at or in an ear canal, respectively, of a user. The BTE part comprises two input transducers (here microphones M.sub.BTE1 and M.sub.BTE2) providing respective (e.g. digitized) electric input signals IN.sub.BTE1 and IN.sub.BTE2 representing sound in the environment. The ITE-part comprises an input transducer (IT2), e.g. a microphone providing, (e.g. digitized) electric input signal IN.sub.ITE representing sound in the environment, and an output unit (OU), e.g. an output transducer, such as a loudspeaker, for providing output stimuli perceivable as sound to the user. The feedback path transfer functions FB1, FB2, FB3 from the output transducer to each of the input transducers (M.sub.BTE1, M.sub.BTE2, IT2, respectively) are indicated together with respective feedback signals v.sub.1, v.sub.2, v.sub.3 and external signals x.sub.1, x.sub.2, x.sub.3 at the location of the three input transducers. The BTE-part further comprises a beamformer unit (BF) receiving the three electric input signals IN.sub.BTE1, IN.sub.BTE2, and IN.sub.ITE representing sound in the environment and providing a beamformed signal IN.sub.BF. The BTE-part further comprises a processor (HLC) for applying a processing algorithm to the beamformed signal, e.g. further noise reduction and/or compressive amplification, etc. and providing a processed electric output signal (OUT), which is fed to the output unit (OU) (in the ITE-part) for presentation to the user. The BTE- and ITE-part are electrically connected via a wired or wireless interface. The BTE-part (here the far-field beamformer filtering unit (BFU)) comprises respective analysis filter banks (t/f) for providing the electric input signals in the frequency domain (e.g. as a number of frequency sub-band signals, e.g. as a ‘map’ of consecutive time-frequency bins (m,k) where m and k are time frame and frequency indices, respectively. Thereby processing of signals can be performed in a time-frequency framework. Similarly, the hearing device, e.g. the BTE-part (and here the processor (HLC)) comprises a synthesis filter bank (t/f) for converting frequency sub-band signals to a time domain signal (OUT) before it is presented to the user via output unit (OU). The far-field beamformer unit (BF) further comprises feedback estimation unit (FBE) for providing estimates (indicated by bold arrow FBEi) of current feedback from the output unit (OU) to at least some (e.g. each) of the input transducers. The feedback estimation unit (FBE) receives the respective electric input signals (IN.sub.BTE1, IN.sub.BTE2, and IN.sub.ITE) and the processed electric output signal (OUT) as inputs for determining the feedback estimates. The far-field beamformer unit (BF) further comprises weighting unit (WGT) for determining weights wij to be applied at a given point in time to the respective electric input signals to properly reflect the current mutual configuration (distances, locations) of ITE and BTE-microphones, cf. discussion above in relation to FIG. 4A. The weights are determined based on the frequency dependent feedback estimates FBEi, which are used to estimate phase (and possibly magnitude) differences between the ITE-microphone and the BTE-microphones (cf. e.g. FIG. 7A, 7B), either adaptively or in advance of use of the hearing device (e.g. during a fitting session where the hearing device is adapted to the user in question).
(58) FIG. 6B shows a second embodiment of a hearing device (HD) comprising a far-field beamformer (BF) according to the second aspect of the present disclosure. The embodiment of FIG. 6B is similar to the embodiment of FIG. 6A, but the beamformer unit (BF) further comprises respective first second and third feedback estimation and cancellation systems (FBE11, FBE12, FBE2) for estimating the respective feedback paths (FB11est, FB12est, FB2est) from the output unit (OU) to each of the input transducers (IT11, IT12, IT2, respectively) and respective subtraction units (′+′) for subtracting the feedback estimates from the respective electric input signals (IN11, IN12, IN2) before they are fed to the beamformer filtering unit (BFU) (cf. signals ERR11, ERR12, ERR2). Thereby the beamformed signal IN.sub.BF provided by the beamformer filtering unit (BF) is based on respective feedback corrected electric input signals (ERR11, ERR12, ERR2).
(59) FIG. 7A shows a difference in magnitude MAG [dB] vs. frequency f [kHz] of a sound signal originating from the output transducer and arriving at the ITE and BTE-microphones, respectively, and FIG. 7B schematically shows a difference in phase PHA [RAD] vs. frequency f [kHz] of a sound signal originating from the output transducer and arriving at the ITE and BTE-microphones, respectively. The magnitude and phase differences are shown relative to the ITE-microphone and represented by the respective curves denoted BTE. FIGS. 7A and 7B illustrate the (shadowing) effect of pinna for propagation of sound from a sound source in the acoustic far-field (approximated by the difference in transfer of sound from an output transducer in the ear canal to each of the ITE and BTE-microphones, which can be derived from estimates of the respective feedback paths, cf. scenario of FIG. 4A). In the sketches of FIGS. 7A and 7B, it is indicated that the effect of pinna is largest between first and second intermediate frequencies f1 and f2, e.g. between two and five kHz (depending on the specific size and form of the ears of the user, hair style, clothing, and possible other ‘wearables’ (e.g. glasses). If the (frequency dependent) differences are adaptively estimated, possible predetermined microphone distances (delay (phase), attenuation (magnitude)) can be (repeatedly) updated (e.g. at each power up of the hearing device, or more frequently, possibly initiated via a user interface) to improve the performance of the far-field beamformer filtering unit (BFU) according to the first and/or second aspect of the present disclosure. In an embodiment, only the phase difference is estimated.
(60) 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.
(61) 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.
(62) 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.
(63) 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.
(64) Accordingly, the scope should be judged in terms of the claims that follow.
REFERENCES
(65) EP2849462A1 (OTICON) 18 Mar. 2015 EP2843971A1 (OTICON) 4 Mar. 2015 EP2701145A1 (RETUNE DSP, OTICON) 26 Apr. 2014 EP3236672A1 (OTICON) 25 Oct. 2017 EP2947898A1 (OTICON) 25 Nov. 2015 EP3185589A1 (OTICON) 28 Jun. 2017
