Feedback cancellation in hearing aid devices using filter tap coherence values

Abstract

The system (20) for hearing assistance includes one or more microphones (23, 24), a speaker (28), and processing circuitry (26). The one or more microphones are mounted near a subject's head and configured to output electrical signals in response to acoustic waves incident on the microphones. The speaker is configured to be mounted near the subject's ears. The processing circuitry amplifies and filters the electrical signals to generate drive signals for input to the speakers using a digital filter (100) having multiple taps, with respective tap coefficients selected to suppress feedback from the speaker to the microphones, and is configured to adaptively calculate tap coefficients, estimating coherence values ​​for each of the tap coefficients over time and weighting updates applied to the tap coefficients in response to the respective coherence values.

Description

[Technical Field]

[0001] The present invention relates generally to hearing aids, and more particularly to devices and methods for acoustic feedback cancellation. [Background technology]

[0002] Speech understanding in noisy environments is a significant challenge for people with hearing impairments. Hearing impairments are typically accompanied by a loss of gain as well as a reduction in the temporal resolution of the sensory system. These characteristics further reduce the hearing-impaired person's ability to filter out target sound sources from background noise, particularly in understanding speech in noisy environments.

[0003] Some newer hearing aids offer a directional hearing mode that improves speech intelligibility in noisy environments. This mode uses an array of microphones and applies beamforming techniques to combine multiple microphone inputs into a single directional audio output channel. The output channel has spatial characteristics that increase the contribution of acoustic waves arriving from a desired direction relative to contributions from other directions.

[0004] For example, PCT International Patent Application Publication No. 2017 / 158507, the disclosure of which is incorporated herein by reference, describes a hearing aid device including a case configured to be physically secured to a mobile phone. An array of microphones are spaced apart within the case and configured to generate electrical signals in response to acoustic input to the microphones. An interface is secured within the case along with processing circuitry coupled to receive and process the electrical signals from the microphones and generate a combined signal for output via the interface.

[0005] As another example, PCT International Patent Application Publication No. 2021 / 074818, the disclosure of which is incorporated herein by reference, describes a hearing aid device including a spectacle frame including a front piece and temples, with one or more microphones mounted at respective first positions on the front piece and configured to output electrical signals in response to first acoustic waves incident on the microphones. A speaker mounted at a second position on one of the temples outputs a second acoustic wave. Processing circuitry processes the electrical signal output by the microphones to generate a drive signal for the speaker, causing the speaker to reproduce a selected sound generated by the first acoustic wave from the first position to the second position with a delay equal to or less than 20% of the transit time of the first acoustic wave, thereby creating constructive interference between the first and second acoustic waves. Summary of the Invention [Problem to be solved by the invention]

[0006] The embodiments of the present invention described below provide improved devices and methods for assisting hearing. [Means for solving the problem]

[0007] One embodiment described herein provides a system for hearing assistance including one or more microphones, a speaker, and processing circuitry. The one or more microphones are mounted near a subject's head and configured to output electrical signals in response to acoustic waves incident on the microphones. The speaker is configured to be mounted near the subject's ears. The processing circuitry amplifies and filters the electrical signals to generate drive signals for input to the speaker using a digital filter having multiple taps, with respective tap coefficients selected to suppress feedback from the speaker to the microphones, and is configured to adaptively calculate tap coefficients, estimating coherence values ​​for each of the tap coefficients over time and weighting updates applied to the tap coefficients in response to the respective coherence values.

[0008] In some embodiments, the processing circuitry is configured to adapt the tap coefficients to estimate a transfer function between the speaker and one or more microphones. In other embodiments, the processing circuitry is configured to adapt the tap coefficients using a gradient descent method with respective convergence factors. In yet other embodiments, the processing circuitry is configured to calculate each convergence factor based on a coherence value.

[0009] In one embodiment, the processing circuitry is configured to calculate the convergence factor by multiplying a common convergence factor by each coherence value. In another embodiment, the processing circuitry is configured to evaluate the coherence value of a given tap based on multiple coefficient updates calculated for the given tap over a specified period of time. In yet another embodiment, a system for providing hearing assistance includes an eyeglass frame, with a microphone and a speaker mounted at respective positions on the eyeglass frame.

[0010] In some embodiments, the one or more microphones include a plurality of microphones, and the processing circuitry is configured to apply a beamforming function to the electrical signals output by the plurality of microphones to emphasize selected sounds occurring within a selected angular range and suppress background sounds occurring outside the selected angular range.

[0011] Further, according to one embodiment described herein, there is provided a method for assisting hearing, comprising mounting an array of microphones near a subject's head, the array of microphones outputting electrical signals in response to acoustic waves incident on the microphones, and mounting speakers near the subject's ears. The electrical signals are amplified and filtered to generate drive signals for input to the speakers using a digital filter having multiple taps, with respective tap coefficients selected to suppress feedback from the speakers to the microphones. The tap coefficients are adaptively calculated, while a coherence value for each of the tap coefficients is estimated over time, and updates applied to the tap coefficients are weighted according to the respective coherence values.

[0012] Further, according to another embodiment described herein, there is provided a head-mounted device (HMD) including a frame, one or more microphones, a speaker, and processing circuitry. The frame is configured to be attached to a subject's head. The one or more microphones are attached to the frame and configured to output electrical signals in response to acoustic waves incident on the microphones. The speaker is attached to the frame. The processing circuitry amplifies and filters the electrical signals to generate drive signals for input to the speaker using a digital filter having multiple taps, with respective tap coefficients selected to suppress feedback from the speaker to the microphones, and is configured to adaptively calculate tap coefficients, estimating coherence values ​​of each of the tap coefficients over time and weighting updates applied to the tap coefficients in response to the respective coherence values.

[0013] In some embodiments, the HMD comprises a device selected from the list comprising: eyewear devices, eyeglasses, eyeglass frames, goggles, helmets, visors, headsets, and clip-on devices. In other embodiments, one or more microphones are attached to the front piece of the frame and speakers are attached to the frame near the subject's ears.

[0014] The present invention will be more fully understood from the following detailed description of the embodiments, when read in conjunction with the drawings. [Brief explanation of the drawings]

[0015] [Figure 1] 1 is a schematic diagram illustrating a hearing aid device based on eyeglass frames, according to one embodiment of the present invention; [Figure 2] 1 is a block diagram that schematically illustrates details of a hearing aid device, according to an embodiment of the present invention; [Figure 3] FIG. 2 is a block diagram that schematically illustrates details of a feedback canceller applicable in a hearing aid device, according to an embodiment of the present invention. [Figure 4A] FIG. 1 is a block diagram that schematically illustrates a processing scheme that supports both beamforming and feedback cancellation, in accordance with an embodiment of the present invention. [Figure 4B] FIG. 1 is a block diagram that schematically illustrates a processing scheme that supports both beamforming and feedback cancellation, in accordance with an embodiment of the present invention. DETAILED DESCRIPTION OF THE INVENTION

[0016] Overview Despite the need for directional hearing aids and the theoretical advantages of microphone arrays in this regard, in practice, the directional performance of hearing aids falls far short of that achieved by natural hearing. Generally, good directional hearing aids require a relatively large number of microphones spaced far apart and designed to be unobtrusive, while still allowing the user to easily aim the hearing aid's directional response toward a point of interest, such as a conversation partner in a noisy environment. Processing circuitry applies beamforming filters to the signals output by the microphones in response to incident acoustic waves to generate an audio output that emphasizes sounds striking the microphone array within an angular range surrounding the direction of interest while suppressing background noise. The audio output should reproduce as natural a hearing experience as possible while minimizing distracting artifacts.

[0017] One of these artifacts is a strong whistling sound that can be caused by acoustic feedback from the audio output of a speaker located near the user's ear to the input of a microphone. Such a whistling sound occurs when the hearing aid's acoustic feedback gain at a given frequency exceeds a certain threshold. Feedback cancellation in hearing aid devices is generally more challenging than in applications such as video conferencing or telephone calls, where echo signals may be delayed by approximately 100 milliseconds. However, in hearing aid devices, feedback signals are typically delayed by less than 20 milliseconds, resulting in a high correlation between the system output and input spectra. Conventional solutions for suppressing or canceling feedback signals include reducing the hearing aid's gain and filtering the audio frequency range in which feedback occurs, but these solutions also reduce the hearing aid's effectiveness in amplifying soft and high-pitched sounds. While it is possible to mechanically reduce feedback gain by placing earmolds on the user's ears, many users find this solution uncomfortable and unsightly.

[0018] Embodiments of the invention described herein address the problem of acoustic feedback by providing a novel method and system for feedback cancellation by estimating a feedback signal and subtracting the feedback signal from an input signal. In the disclosed embodiment, an array of microphones mounted near a user's head outputs electrical signals in response to incoming acoustic waves incident on the microphones. A speaker is mounted near the user's ear. Processing circuitry amplifies and filters the electrical signals to generate drive signals for input to the speaker using a digital filter having multiple taps, with respective tap coefficients selected to suppress feedback from the speaker to the microphone, and adaptively calculates tap coefficients, estimating coherence values ​​for each of the tap coefficients over time and weighting updates applied to the tap coefficients according to the respective coherence values.

[0019] In some embodiments, the microphone and speaker are mounted to a frame attached to the user's head. In some of the embodiments described below, the microphone and speaker are mounted to eyeglass frames. Alternatively, the microphone and speaker can be mounted to other types of frames or head-mounted devices (HMDs), such as virtual reality (VR) or augmented reality (AR) headsets, or in other types of mounting configurations.

[0020] In the present context, an HMD comprises any type of frame to which a microphone and a speaker can be attached. The HMD may be selected from a list comprising, but not limited to, eyewear devices, eyeglasses, eyeglass frames, goggles, helmets, visors, headsets, and clip-on devices. In some embodiments, one or more microphones are attached to the front piece of the frame and speakers are attached to the frame near the subject's ears.

[0021] In some embodiments, the processing circuitry adapts the tap coefficients to estimate a transfer function between the speaker and one or more microphones. The processing circuitry uses the estimated transfer function to estimate a feedback signal to be subtracted from the input signal. The processing circuitry may adapt the tap coefficients using a gradient descent method with respective convergence coefficients, which the processing circuitry calculates based on the coherence values. In one embodiment, the processing circuitry calculates the convergence coefficients by multiplying a common convergence coefficient by the respective coherence values.

[0022] In some embodiments, the processing circuitry evaluates the coherence value for a given tap based on multiple coefficient updates calculated for the given tap over a specified period of time.

[0023] In some embodiments, the system comprises an eyeglass frame, with the microphone and speaker mounted at respective positions on the eyeglass frame.

[0024] In one embodiment, the one or more microphones comprise a plurality of microphones, and the processing circuitry applies a beamforming function to the electrical signals output by the plurality of microphones to emphasize selected sounds occurring within a selected angular range and to suppress background sounds occurring outside the selected angular range.

[0025] System Description FIG. 1 is a schematic diagram of a hearing aid device 20 integrated into an eyeglass frame 22, according to one embodiment of the present invention. An array of microphones 23, 24 is mounted at respective locations on the eyeglass frame 22 and outputs electrical signals in response to acoustic waves incident on the microphones. In the illustrated embodiment, microphone 23 is mounted on a front piece 30 of the frame 22, while microphone 24 is mounted on temples 32 connected to each end of the front piece 30. While the large array of microphones 23, 24 illustrated in FIG. 1 is useful in some applications of the present invention, the signal processing and hearing aid principles described herein may alternatively be applied, mutatis mutandis, using fewer microphones. For example, these principles may be applied using an array of microphones 23 on the front piece 30, as well as to devices using other microphone mounting configurations that are not necessarily eyeglass-based.

[0026] Processing circuitry 26 is fixed within or connected to eyeglass frame 22 and coupled by electrical wiring 27, such as traces on a flexible printed circuit, to receive the electrical signals output from microphones 23 and 24. While processing circuitry 26 is illustrated in FIG. 1 , for simplicity, some or all of the processing circuitry may instead be located in front piece 30 or in a unit externally connected to frame 22 at certain locations in temple 32. Processing circuitry 26 generates audio output with a particular directional response by mixing the signals from the microphones and applying, for example, a beamforming function to emphasize sounds occurring within a selected angular range and suppress background sounds occurring outside that range. Typically, though not necessarily, the directional response coincides with the angular orientation of frame 22. The processing circuitry additionally suppresses the acoustic signals picked up by the microphones and emitted by the speakers.

[0027] These signal processing functions of processing circuitry 26 are described in further detail below.

[0028] The processing circuitry 26 may transmit audio output to the user's ears via any suitable type of interface and speaker. In the illustrated embodiment, the audio output is generated by a drive signal to drive one or more audio speakers 28 mounted in temples 32, typically near the user's ears. While only one speaker 28 is shown in FIG. 1 on each temple 32, the device 20 may instead include only a single speaker on one of the temples 32, or two or more speakers mounted on one or both temples 32. In the latter case, the processing circuitry 26 may apply a beamforming function to the drive signal to direct acoustic waves from the speakers toward the user's ears. Alternatively, the drive signal may be transmitted to a speaker inserted in the ear, or may be transmitted via a wireless connection, e.g., as a magnetic signal, to a telecoil in a user's hearing aid (not shown) wearing the eyeglass frames.

[0029] Signal Processing Figure 2 is a block diagram that schematically illustrates details of the processing circuitry 26 in the hearing aid device 20 according to an embodiment of the present invention. The processing circuitry 26 may be implemented in a single integrated circuit chip, or alternatively, the functions of the processing circuitry 26 may be distributed across multiple chips that may be located inside or outside the eyeglass frame 22. Although one particular implementation is illustrated in Figure 2, the processing circuitry 26 may comprise any suitable combination of analog and digital hardware circuits, along with appropriate interfaces for receiving electrical signals output by the microphones 23, 24 and outputting drive signals to the speaker 28.

[0030] In this embodiment, the microphones 23, 24 include integrated analog-to-digital converters and output digital audio signals to the processing circuitry 26. Alternatively, the processing circuitry 26 may include an analog-to-digital converter that converts the analog outputs of the microphones to digital form. The processing circuitry 26 typically includes suitable programmable logic components 40, such as a digital signal processor (DSP) or gate array, to perform the necessary filtering, mixing, and feedback cancellation functions to generate and output drive signals in digital form for the speaker 28.

[0031] These filtering and mixing functions typically involve the application of beamforming filters 42 having coefficients selected to create a desired directional response. Specifically, in some embodiments, the coefficients of the beamforming filters 42 are calculated to emphasize sounds that strike the frame 22 (and thus the microphones 23, 24) within a selected angular range. Details of filters that may be used for beamforming purposes are described further below.

[0032] Alternatively or additionally, processing circuitry 26 may comprise a neural network (not shown) trained to determine and apply coefficients used in beamforming filter 42. Further alternatively or additionally, processing circuitry 26 comprises a microprocessor programmed in software or firmware to perform at least some of the functions described herein.

[0033] Processing circuitry 26 may apply any suitable beamforming function known in the art, either in the time domain or the frequency domain, in implementing beamforming filter 42. Beamforming algorithms that may be used in this context are described, for example, in the above-mentioned PCT International Patent Application Publication No. 2017 / 158507 (especially pages 10-11) and U.S. Pat. No. 10,567,888 (especially paragraph 9).

[0034] In one embodiment, processing circuitry 26 applies a minimum variance distortion-free response (MVDR) beamforming algorithm in deriving the coefficients of beamforming filter 42. This type of algorithm is advantageous in achieving fine spatial resolution and distinguishing between sounds originating from a direction of interest and sounds originating from the user's own voice. The MVDR algorithm maximizes the signal-to-noise ratio (SNR) of the audio output by minimizing the average energy (while keeping the target distortion small). This algorithm can be implemented in frequency space by calculating a complex weight vector F(ω) of the output signal from each microphone at each frequency, expressed by the following equation:

[0035]

number

[0036] In an alternative embodiment, processing circuitry 26 applies a linearly constrained minimum variance (LCMV) algorithm in deriving the coefficients of beamforming filter 42. LCMV beamforming forces the beamforming filter to pass signals from the desired direction with a specified gain and phase delay while minimizing power from interfering signals and noise from all other directions.

[0037] In some embodiments, processing circuitry 26 includes a feedback canceller 44 that suppresses acoustic feedback from the speaker to the microphone. To this end, feedback canceller 44 adaptively calculates tap coefficients using a digital filter (not shown) having multiple taps, with respective tap coefficients selected to suppress feedback from the speaker to the microphone, estimating coherence values ​​for each of the tap coefficients over time and weighting updates applied to the tap coefficients according to the respective coherence values. Feedback cancellers are described in more detail below with reference to FIG. 3.

[0038] Audio output circuitry 46, including, for example, an appropriate codec and digital-to-analog converter, converts the digital drive signal output from beamforming filter 42 (or from feedback canceller 44 following the beamforming filter) to analog form. Analog filter 48 further performs filtering and analog amplification functions to optimize the analog drive signal to speaker 28.

[0039] A control circuit 50, such as an embedded microcontroller, controls the programmable functions and parameters of the processing circuitry 26, which may include the feedback canceller 44. A communications interface 52, such as a Bluetooth® or other wireless interface, allows a user and / or hearing professional to set and adjust these parameters as needed. A power circuit 54, such as a battery inserted in the temple 32, provides power to the other components of the processing circuitry.

[0040] Feedback Cancellation Processing As mentioned above, sound waves generated by the speaker of a hearing aid device can be picked up by the microphone of the hearing aid device, creating a whistling or howling sound. The purpose of a feedback canceller is to prevent whistling artifacts by reducing the amount of feedback signal in the signal generated by the microphone.

[0041] Next, the principle of feedback cancellation will be explained. Let Out(t) represent the signal output by the hearing aid device, p(t) represent the signal received at the microphone from the output of the hearing aid device only (the version of Out(t) received by the microphone), and y(t) represent the signal received by the microphone from all audio sources other than the speaker of the hearing aid device, where t represents the time axis. The overall signal x(t) generated by the microphone is given by x(t) = y(t) + p(t).

[0042] The feedback canceller generates a feedback signal based on an output signal Out(t-Δt) (e.g., a reference signal) generated a period Δt before, as follows:

number

number

number

[0043] The feedback canceller then subtracts the estimated feedback signal from x(t) to obtain

number

number

[0044] 3 is a block diagram that schematically illustrates details of a feedback canceller 44 that can be applied to a hearing aid device 20 according to an embodiment of the present invention. Alternatively, the principles of this feedback canceller can be applied to other devices and systems with appropriate microphone arrays, speakers, and signal processing capabilities.

[0045] The feedback canceller 44 of Figure 3 implements the feedback cancellation principle described above in a digital form, where various signals are sampled on a digital time axis, denoted 'n'. In the embodiment of Figure 3, the feedback canceller 44 receives an input signal x(n) received by microphones 23, 24 and including the feedback signal from speaker 28. The feedback canceller uses subtractor 104 to derive an estimated feedback signal x(n) from x(n).

[0046]

number

[0047] The feedback canceller 44 is an adaptive filter having N taps with respective tap coefficients.

number

[0048] The tap adapter 108 updates the tap coefficients of the adaptive filter 100 using any suitable gradient descent method, such as the LMS or NLMS method. Let Δh(n) represent a vector of coefficient updates each corresponding to a tap of the adaptive filter 100. The vector Δh(n) has the same length N as the adaptive filter 100. In this embodiment, the tap adapter

[0049]

number

[0050] Here, the update vector Δh(n) is Δh(n)=μ Out(n) X(n) is given by

[0051] μ is the scalar convergence factor of the underlying gradient descent method, and the vector X(n) is X(n)=[x(n-N+1)…x(n)] is given by

[0052] Next, embodiments are described in which the adaptation of tap coefficients by the tap adapter 108 is based on multiple convergence coefficients rather than a single scalar convergence coefficient. In such embodiments, a common convergence coefficient μ is weighted by a respective weight value for each tap. In some embodiments, the tap adapter calculates the weight values ​​by calculating the respective tap coherence values, as described herein. This approach provides a time-based weighting mechanism for modifying the updates Δh(n) applied to the tap coefficients of the adaptive filter. The inventors have discovered that, for example, in open-ear hearing aid eyewear, weighting the updates of the tap coefficients with the respective time coherence values ​​of the taps can significantly improve feedback cancellation performance.

[0053] The performance of the feedback cancellation method can be determined, for example, by measuring the maximum acoustic output gain at which the underlying system remains stable without whistling. The inventors have found that the applicable gain using the disclosed coherence-based feedback cancellation method is significantly higher than the gain achievable if the coherence value is omitted.

[0054] In general, using a coherence value involves evaluating the updates adaptively applied to each tap of the adaptive filter over a short period, e.g., 16 milliseconds (or any other suitable period), and weighting each tap coefficient update based on the coherence value. In some embodiments, the coherence value C for weighting the coefficient update for the ith tap is i For example,

[0055]

number

[0056] where n denotes the digital time index,

number

[0057]

number

number

[0058] Coherence value C i indicates the respective confidence level associated with the coefficient update. The slope coefficient μ has a higher weight when the tap is associated with a large coherence value (considering the update to be more reliable) and a lower weight when the tap is associated with a small coherence value (considering the update to be less reliable).

[0059] The above method for calculating the coherence value is given as an example and other types of coherence values ​​can also be used. For example, a reduced complexity code coherence value can be

number

[0060] As another example, the coherence values ​​are: i To,

number

[0061] where the sum of the last equation is taken over the number of taps W'>1.

[0062] In embodiments where feedback cancellation is performed in the frequency domain, a phase coherence factor can be applied. An exemplary formulation of this type can be found, for example, in the paper entitled “Phase Coherence Imaging: Principles, applications and current developments,” Bruges, Belgium, Signal Processing in Acoustics: PSP (2 / 3) Presentation 1.

[0063] Beamforming and Feedback Cancellation Schemes 4A and 4B are block diagrams that schematically illustrate a processing scheme that supports both beamforming and feedback cancellation, in accordance with an embodiment of the present invention.

[0064] The schemes in FIG. 4A and FIG. 4B differ from each other in the order in which beamforming and feedback cancellation are performed.

[0065] In the scheme of Figure 4A, input signals from multiple microphones are processed by a beamforming filter, such as, for example, beamforming filter 42 of Figure 2 above. The signals output by the beamforming filter are then feedback canceled, using, for example, feedback canceller 44 of Figures 2 and 3. In such an embodiment, the adaptive filter of Figure 3

[0066]

number

[0067] In the scheme of Figure 4B, input signals from multiple microphones are each processed by a dedicated feedback canceller 44. The output of the feedback canceller is input to a beamforming filter 42, the output of which is provided to the loudspeaker via interface 120. In such an embodiment, the adaptive filter of Figure 3

[0068]

number

[0069] The scheme of Figure 4A is less complex than the scheme of Figure 4B because it has only one feedback canceller instead of multiple feedback cancellers. Furthermore, the beamforming performance in the scheme of Figure 4A may be better than that of Figure 4B because applying separate feedback cancellation to each microphone (as in Figure 4B) may reduce the correlation between microphones, which is necessary for proper operation of the beamforming filter.

[0070] On the other hand, the scheme in Figure 4B may be advantageous over that of Figure 4A because more information and degrees of freedom are available to mitigate problems such as feedback when performing feedback cancellation at each microphone individually. Furthermore, providing multiple microphone feedback-free audio channels (as in Figure 4B) can be used in implementing various algorithms other than beamforming. Exemplary related algorithms in this regard include (but are not limited to) own voice detection, direction of arrival estimation, transfer function, and room sound level measurement.

[0071] Although the embodiments described herein primarily address feedback cancellation in hearing aid devices, the methods and systems described herein can also be used for feedback cancellation in other HMD devices and in other applications such as noise-canceling headphones.

[0072] The above-described embodiments are given by way of example, and it will be recognized that the scope of the following claims is not limited to what is particularly shown and described above. Rather, the scope includes both combinations and subcombinations of the various features described above, as well as variations and modifications thereof not disclosed in the prior art that would occur to one skilled in the art upon reading the foregoing description. Documents incorporated by reference into this patent application shall be deemed an integral part of this application, except that, for terms defined in such incorporated documents in a manner that contradicts a definition expressly or implicitly given herein, only the definition given herein shall be considered.

Claims

1. 1. A system for a hearing aid, comprising: one or more microphones mounted near the subject's head and configured to output electrical signals in response to acoustic waves incident on the microphones; a speaker configured to be mounted near the subject's ear; and processing circuitry configured to amplify and filter the electrical signal to generate a drive signal for input to the speaker using a digital filter having a plurality of taps, the digital filter having respective tap coefficients selected to suppress feedback from the speaker to the microphone, and configured to adaptively calculate the tap coefficients while estimating a coherence value for each of the tap coefficients over time and weighting updates applied to the tap coefficients in response to the respective coherence value.

2. The system of claim 1 , wherein the processing circuitry is configured to adapt the tap coefficients to estimate a transfer function between the speaker and the one or more microphones.

3. The system of claim 1 , wherein the processing circuitry is configured to adapt the tap coefficients using a gradient descent method having respective convergence factors.

4. The system of claim 3 , wherein the processing circuitry is configured to calculate the convergence factors respectively based on the coherence values.

5. The system of claim 3 , wherein the processing circuitry is configured to calculate the convergence factors by multiplying a common convergence factor by the respective coherence values.

6. 4. The system of claim 3, wherein the processing circuitry is configured to estimate a coherence value for a given tap based on multiple coefficient updates calculated for the given tap over a specified period of time.

7. The system according to any one of claims 1 to 6, comprising an eyeglass frame, the microphone and the speaker being mounted at respective positions on the eyeglass frame.

8. 7. The system of claim 1, wherein the one or more microphones comprise a plurality of microphones, and wherein the processing circuitry is configured to apply a beamforming function to the electrical signals output by the plurality of microphones to emphasize selected sounds occurring within a selected angular range and suppress background sounds occurring outside the selected angular range.

9. 1. A method for a hearing aid, comprising: mounting an array of microphones near the subject's head, the array outputting electrical signals in response to acoustic waves incident on the microphones; Mounting a speaker near the subject's ear; amplifying and filtering the electrical signal to generate a drive signal for input to the speaker using a digital filter having a plurality of taps, the tap coefficients selected to suppress feedback from the speaker to the microphone; and adaptively calculating the tap coefficients while estimating a coherence value for each of the tap coefficients over time and weighting updates applied to the tap coefficients in response to the respective coherence values.

10. The method of claim 9 , wherein calculating the tap coefficients comprises adapting the tap coefficients to estimate a transfer function between the speaker and the one or more microphones.

11. The method of claim 9 , wherein calculating the tap coefficients comprises adapting the tap coefficients using a gradient descent method having respective convergence factors.

12. The method of claim 11 , comprising calculating each of the convergence factors based on the coherence value.

13. The method of claim 11 , wherein calculating the convergence factor comprises multiplying the respective coherence values ​​by a common convergence factor.

14. 12. The method of claim 11, comprising evaluating a coherence value for a given tap based on multiple coefficient updates calculated for the given tap over a specified period of time.

15. The method according to any one of claims 9 to 14, wherein the microphone and the speaker are mounted at respective positions on an eyeglass frame.

16. 15. The method of claim 9, wherein the one or more microphones comprise a plurality of microphones, and the method comprises applying a beamforming function to the electrical signals output by the plurality of microphones to emphasize selected sounds occurring within a selected angular range and suppress background sounds occurring outside the selected angular range.

17. A head-mounted device (HMD), a frame configured to be attached to the subject's head; and one or more microphones mounted to the frame and configured to output electrical signals in response to acoustic waves incident on the microphones; a speaker attached to the frame; a processing circuit configured to amplify and filter the electrical signal to generate a drive signal for input to the speaker using a digital filter having a plurality of taps, the digital filter having respective tap coefficients selected to suppress feedback from the speaker to the microphone; and a processing circuit configured to adaptively calculate the tap coefficients while estimating a coherence value of each of the tap coefficients over time and weighting updates applied to the tap coefficients according to the respective coherence values.

18. 20. The HMD of claim 17, wherein the HMD comprises a device selected from the list comprising an eyewear device, eyeglasses, eyeglass frames, goggles, helmets, visors, headsets, and clip-on devices.

19. 19. An HMD according to claim 17 or 18, wherein the one or more microphones are mounted on a front piece of the frame and the speakers are mounted on the frame near the subject's ears.

Images