Abstract: A method for operating a hearing aid which has an auditory canal microphone for capturing sound in the auditory canal of the wearer and an own voice recognition unit for carrying out an adaptive own voice recognition. Sound in the auditory canal is captured by the auditory canal microphone and a signal that corresponds to the sound is output by the auditory canal microphone, and training of the adaptive own voice recognition is controlled only based on the signal output by the auditory canal microphone. A corresponding hearing aid configured to carry out the method.

Abstract: An earmold having an earmold shell, the earmold shell having a first end facing a tympanic membrane when the earmold is worn by a user, and a second end facing toward a surrounding of the user when the earmold is worn by the user, includes: a receiver configured to provide an audio output signal; a receiver channel coupled to an output of the receiver and extending to a receiver opening in the first end; and a vent channel coupled to the receiver channel through a first vent port, the vent channel having a vent opening in the second end; wherein the receiver channel comprises a closing element, the closing element comprising a first magnetic member, wherein the closing element is configured to cause the first vent port to be open when in a first state, and to cause the first vent port to be closed when in a second state.

Abstract: A hearing device includes a feedback control system that applies an adaptive filtering algorithm. The adaptive algorithm provides a filter control signal to adaptively control filter coefficients based on first and second algorithm input signals of a forward path. The feedback control system further includes first and second transform units for transforming the first and second algorithm input signals to the transform domain, and an inverse transform unit to convert an estimate of the current feedback path in the transformed domain to a time domain estimate, and a combination unit in the forward path to subtract the estimate of the current feedback signal from a signal of the forward path to provide a feedback corrected signal.

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.

Abstract: An audio system for artificial reality systems is configured to reduce feedback and mitigate entrainment in audio content presented to the user. The audio system utilizes adaptive feedback cancellation processes to reduce feedback. The audio system processes reference signals in a hybrid of the time domain and the frequency domain to achieve rapid convergence with decreased entrainment. The audio system may pre-whiten reference signals and speaker signals and separate the pre-whitened signals into frequency bands using a filter bank. The audio system uses a state space algorithm to generate adaptive filters in each frequency band. The adaptive filters may be applied to the speaker signal to generate a feedback cancelation signal which may be subtracted from the reference signal to decrease feedback.

Abstract: A whistling sound suppression method includes: obtaining an ambient audio signal, the ambient audio signal being a sound signal in a surrounding environment of an earphone; filtering the ambient audio signal according to a preset first filter group to obtain a first audio signal; obtaining an ear canal audio signal, the ear canal audio signal being a sound signal when the first audio signal propagates in an ear canal; and filtering a subsequently obtained ambient audio signal according to a preset second filter group to obtain a second audio signal in response to the ear canal audio signal meets a whistling condition.

Abstract: A method of determining an audio controller for a headphone that is configured to use an acoustic transducer to develop sound that is delivered to an ear of a user and that includes a feedback microphone that is configured to sense sound developed by the acoustic transducer, and a related computer program product and system. A first audio transfer function between the acoustic transducer and the feedback microphone is measured. A second audio transfer function between the acoustic transducer and the feedback microphone with a feedback controller applied is determined. The audio controller is calculated based on both the first audio transfer function and the second audio transfer function.

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

Abstract: The present disclosure relates to a bone conduction speaker and its compound vibration device. The compound vibration device comprises a vibration conductive plate and a vibration board, the vibration conductive plate is set to be the first torus, where at least two first rods inside it converge to its center; the vibration board is set as the second torus, where at least two second rods inside it converge to its center. The vibration conductive plate is fixed with the vibration board; the first torus is fixed on a magnetic system, and the second torus comprises a fixed voice coil, which is driven by the magnetic system. The bone conduction speaker in the present disclosure and its compound vibration device adopt the fixed vibration conductive plate and vibration board, making the technique simpler with a lower cost; because the two adjustable parts in the compound vibration device can adjust both low frequency and high frequency area, the frequency response obtained is flatter and the sound is broader.

Abstract: In one example, a method includes determining, by one or more processors of a hearing aid programmer, values for one or more properties of a first hearing aid; obtaining, by the one or more processors, based on feedback cancelation configurations of a plurality of other hearing aids having the same values for the one or more properties as the first hearing aid, a predicted initial feedback cancelation configuration for the first hearing aid; and programming, by the one or more processors, the first hearing aid based on the predicted initial feedback cancelation configuration.

Abstract: A method of defining and setting a nonlinear signal processing of a hearing device, e.g. a hearing aid, by machine learning is provided.

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.

Abstract: A method of focused wireless power transmission is disclosed which includes generating a map for electromagnetic fingerprints at N locations within an environment of interest, including a transmitter and plurality of receivers each located at one of the N locations, transmitting pre-matched signals based on the electromagnetic fingerprints in a predetermined order for a first location of the N location, measuring response at each of the N locations, measuring spatial correlations between all other locations and the first location, and evaluating electromagnetic power focus at each of the N locations by comparing the measured spatial correlations.

Abstract: A hearing system includes first and second hearing devices adapted for being located at first and second ears of a user, or for being fully or partially implanted in the head at the left and right ears of the user. The first hearing device includes a forward path having a1) an input transducer for converting a sound at the first hearing device to a first electric input signal including the sound; a processor for processing the first electric input signal, or a signal originating therefrom, and providing a first processed signal in dependence of a reduced hearing ability of the user at the first ear; an output unit adapted for providing stimuli perceivable as sound for the user at the first ear based on the first processed signal.

Abstract: Techniques for improving adaptive interference cancellation (AIC) using cascaded AIC algorithms are described. To improve an accuracy of detecting speech, a device may perform a first stage of AIC to generate isolated audio data and may generate speech mask data indicating time windows when speech is detected in the isolated audio data. Based on the speech mask data, the device may perform second AIC to generate output audio data, with adaptation of the adaptive filter enabled when the speech is not detected and disabled when the speech is detected. Thus, the first AIC improves the accuracy with which the device detects that speech is present and the second AIC reduces distortion in the output audio data by not updating filter coefficient values when the speech is present. The first AIC may use playback audio data, microphone audio data or beamformed audio data as reference signals.

Abstract: An in-ear device includes a tympanic membrane measurement unit (TMMU) and an audio package configured to emit sound. The device is configured to measure a movement of a tympanic membrane in an ear using the TMMU caused by external sound received by the tympanic membrane, analyze at least the movement of the tympanic membrane measured by the TMMU using a predictive model trained by a machine learning technique to generate a waveform of cancellation sound that will destructively interfere with the external sound when received by the tympanic membrane, and output the cancellation sound from the audio package to destructively interfere with external sound received by the tympanic membrane.

Abstract: A speech intelligibility enhancing system for difficult acoustical conditions is disclosed, the speech intelligibility enhancing system comprising at least one ear plug (201) for insertion in an ear canal (218) of a person, the at least one ear plug being arranged with an ear canal facing portion (401) and an environment facing portion (402), and the at least one ear plug comprising an acoustically attenuating path (214; 214, 213) comprising a vent (214) coupling said environment facing portion (402) with said ear canal facing portion (401); and an electroacoustic path (202, 204, 209; 202, 203, 204, 208, 209, 210, 211, 212) comprising a microphone (202) at said environment facing portion (402), a variable gain (204) and a loudspeaker (209) at said ear canal facing portion (401); wherein said acoustically attenuating path (214; 214, 213) is arranged with a transfer function from said environment facing portion (402) to said ear canal facing portion (401) having a low pass characteristic having a low pass cut¬o

Abstract: A method performed by an audio system that includes a headset with a left headset housing and a right headset housing. The method includes driving a speaker of the left headset housing with an audio signal, determining whether audio howl is present within the left headset housing by comparing spectral content from a first error microphone signal produced by a first error microphone of the left headset housing and spectral content from a second error microphone signal produced by a second error microphone of the right headset housing, and, in response to determining that audio howl is present, filtering the audio signal to mitigate the audio howl.

Abstract: A hearing device comprises an input transducer providing an input gain GI, a signal processor comprising a compressor for determining a frequency and level dependent desired compressor gain GP to compensate for a hearing impairment of the user, and to provide a resulting compressor gain G?P, and an output transducer for providing output stimuli perceivable as sound for the user based on a processed signal, the output transducer providing an output gain, GO. A resulting forward path gain G? is defined in a logarithmic representation as GI+G?P+GO. The hearing device further comprises a loop gain estimator for continuously estimating a current loop gain ?L(n), configured to provide a loop gain estimate within a predefined number of feedback loop delays after a feedback buildup has started, and a loop gain controller for dynamically controlling said resulting forward path gain G? in dependence of said estimate of said current loop gain ?L(n).

Abstract: A hearing aid system (500) with active noise cancelling and a method for operating such a hearing aid system.

Abstract: A method of conducting a fitting session for fitting a hearing device to a hearing device user's needs is provided. The method comprises S1. providing an estimate of, a current feedback from said output transducer to said input transducer, while the hearing device is in an operational state; S2. evaluating said estimate of a current feedback and providing a value of a feedback risk indicator in dependence of said estimate of a current feedback; S3. determining whether said value of the feedback risk indicator fulfils a high-risk criterion; and S4. if said high-risk criterion is fulfilled providing at least one of a warning, a recommendation, and an action in relation to said feedback risk. Steps S1 to S4 are configured to be automatically performed as background processes. Thereby a simplified scheme for fitting a hearing device to a user's needs may be provided.

Abstract: The present disclosure provides a hearing protection system and a method for estimating a voice signal of a hearing protection system user. The hearing protection system comprises an ear canal microphone for provision of an ear canal input signal; a receiver for provision of an audio output signal; a compensation module for receiving and filtering the ear canal output signal for provision of a compensation signal; and a mixer connected to the ear canal microphone and the compensation module for provision of a voice signal, wherein the compensation module comprises a filter controller, a primary filter and a secondary filter, wherein the primary filter is a static filter, wherein primary filter coefficients of the primary filter are static, and wherein the secondary filter is an adaptive filter, wherein secondary filter coefficients of the secondary filter are controlled by the filter controller based on the voice signal.

Abstract: A hearing device, e.g. a hearing aid, comprises a) a multitude of input transducers for providing respective electric input signals representing sound in an environment of the user; b) an output transducer for providing stimuli perceivable to the user as sound based on said electric input signals or a processed version thereof; c) an adaptive beamformer filtering unit connected to said input unit and to said output unit, and configured to provide a spatially filtered signal based on said multitude of electric input signals and adaptively updated beamformer weights; and d) a feedback estimation unit providing feedback estimates of current feedback paths from said output transducer to each of said input transducers. The hearing device is configured to provide that at least one of said adaptively updated beamformer weights of the adaptive beamformer filtering unit is/are updated in dependence of said feedback path estimates. The application further relates to a method of suppressing feedback.

Abstract: An external portion of an auditory prosthesis includes an external magnet that interacts with an implantable magnet to hold the external portion against the skin. Magnetic force generated by the stray field of these magnets can disturb the operation of a vibrating element of the auditory prosthesis. The technologies described herein utilize additional magnets disposed within portions of the auditory prosthesis to redirect the magnetic flux, which allows the vibrating element to be disposed more closely to the magnets, reducing the overall height profile of the prosthesis.

Abstract: A diaphragm with a hinge portion and a drive portion and a plurality of oblong frame portions between the hinge portion and drive portion. The oblong frame portions are able to vibrate independently of each other and may have different resonance frequencies.

Abstract: A hearing device, e.g. a hearing aid, comprises a) an input unit comprising a multitude of input transducers for providing respective electric input signals representing sound in an environment of the user; b) an output unit comprising an output transducer for providing stimuli perceivable to the user as sound based on said electric input signals or a processed version thereof; c) a spatial filter connected to said input unit and to said output unit, and configured to provide a spatially filtered signal based on said multitude of electric input signals and configurable beamformer weights; d) a spatial filter controller configured to apply first and/or second different sets of beamformer weights to said multitude of electric input signals, wherein said first set of beamformer weights is applied to provide spatial filtering of sound from said output transducer, and wherein said second set of beamformer weights is applied to provide spatial filtering of an external sound field.

Abstract: A hearing device configured to be fitted at or in a user's ear canal including an acoustic vent configured to enable sound waves to pass through the hearing device. A directional microphone is configured to create an output signal by amplifying sound traveling in a first direction through the acoustic vent toward the ear canal and attenuating sound traveling in a second direction through the acoustic vent from the ear canal. A receiver is configured to produce sound in response to the output signal. A method of operating a hearing device is also included.

Abstract: Presented herein are techniques for suppressing electrical interference caused by a feedback path(s) between an output channel and an input channel of an implantable medical device. In particular, the output channel carries stimulation signals that can pass through the feedback path(s) and appear in the input channel, thereby causing interference with any input signals carried by the input channel. The implantable medical device is configured to determine characteristics of the feedback and subsequently use the feedback characteristics to generate feedback compensation signals for application to the input channel.

Abstract: An in-ear device includes a housing shaped to hold the in-ear device in an ear, and an audio package, disposed in the housing, to emit sound. A tympanic membrane measurement unit (TMMU) is structured to measure a movement of a tympanic membrane in the ear caused by external sound received by the tympanic membrane, and a controller is coupled to the audio package and the TMMU. The controller includes logic that when executed by the controller causes the in-ear device to perform operations. The operations include measuring a movement of the tympanic membrane, and in response to measuring the movement of the tympanic membrane, outputting sound from the audio package to destructively interfere with the external sound received by the tympanic membrane.

Abstract: A hearing aid (1) comprises a flexible ear tip (9) for an ear piece (8) adapted for insertion into the ear canal of a user. The ear piece comprises a shroud having a generally ball-shaped hollow bulb made of soft resilient material, featuring a mid zone that is adapted for being softer to compression by the contact with ear canal wall than is the case with the end zones. The invention further provides an ear piece, an ear plug, and a component for a communications device.

Abstract: The present disclosure pertains to hearing devices configurable between open fit and closed fit configurations at different times through actuation of one or more acoustic valves located in one or more corresponding sound passages of the hearing device. The one or more acoustic valves of the hearing device are adaptively controlled based on context detected by one or more sensors. The context may be, but is not limited to, a mode of operation of the hearing devices which may include, for example, an audio content playback mode and a voice communication mode. The actuatable valves may be actuatable in situ without having to remove the hearing device from the user's ear thereby enabling the user to experience the benefit of a closed fit or an open fit depending on the user's desire or other context.

Abstract: A hearing device, e.g. a hearing aid, comprises a forward path for processing an electric signal representing sound including a) an input unit for receiving or providing an electric input signal representing sound, b) a signal processing unit, c) an output transducer for generating stimuli perceivable as sound to a user, d) a feedback detection unit configured to detect feedback or evaluate a risk of feedback via an acoustic or mechanical or electrical feedback path from said output transducer to said input unit and comprising d1) a magnitude and phase analysis unit for repeatedly determining magnitude, Mag, and phase, Phase, of said electric input signal and further parameters based thereon, and d2) a feedback conditions and detection unit configured to check criteria for magnitude and phase feedback condition, respectively, based on said values, and to provide a feedback detection signal indicative of feedback or a risk of feedback.

Abstract: A hearing device includes: a first input transducer configured to convert an acoustic signal into a first input signal; a processing unit configured to provide a processed signal based on the first input signal; an acoustic output transducer configured to provide an audio output signal of based on the processed signal; a second input transducer configured to provide a second input signal based at least on the audio output signal from the acoustic output transducer and a body-conducted voice signal from a user of the hearing device; and a user voice extraction unit connected to the processing unit for receiving the processed signal, and connected to the second input transducer for receiving the second input signal, wherein the user voice extraction unit is configured to extract a voice signal based at least on the second input signal and the processed signal.

Abstract: An audio system includes a transducer assembly, an optical sensing pathway, a laser, a detector assembly, and a controller. The transducer assembly is coupled to a user's ear and produces an acoustic pressure wave based on an audio instruction. The optical sensing pathway moves, at least in part, with a detected acoustic pressure wave. The laser emits light that is separated into a reference beam and a sensing beam that is coupled into the optical sensing pathway. The detected acoustic pressure wave interacts with the sensing beam to alter its optical path length. The detector assembly detects the reference and sensing beams from the optical sensing pathway, and measures the detected acoustic pressure wave based on changes in optical path length between the reference beam and the sensing beam. The controller adjusts the audio instruction based on the measurement of the detected acoustic pressure wave.

Abstract: A hearing aid includes a signal processor, a input transducer electrically connected to the signal processor, a receiver electrically connected to the signal processor, an adaptive feedback cancellation filter configured to suppress feedback from a signal path between the receiver and the input transducer, and a feedback gain correction unit configured for adjusting a gain parameter of the signal processor based at least in part on coefficients of the adaptive feedback cancellation filter. A method of adjusting a gain parameter of a signal processor of a hearing aid includes monitoring filter coefficients of a feedback cancellation filter of the hearing aid, and adjusting the gain parameter of the signal processor in dependence of the monitored filter coefficients.

Abstract: In one example, a method includes determining, by one or more processors of a hearing aid programmer, values for one or more properties of a first hearing aid; obtaining, by the one or more processors, based on feedback cancelation configurations of a plurality of other hearing aids having the same values for the one or more properties as the first hearing aid, a predicted initial feedback cancelation configuration for the first hearing aid; and programming, by the one or more processors, the first hearing aid based on the predicted initial feedback cancelation configuration.

Abstract: A digital signal processor processes acoustic sound in a body-worn hearing assist device including separating the microphone signal into frequency bands. Two of the frequency bands, which are preferably adjacent, are considered as a pair. In one of the frequency bands in the pair, the band signal is replicated/split into two subsignals. One of the subsignals is frequency shifted into the other frequency band of the pair. The input signal from the paired frequency band is either significantly attenuated or altogether discarded. The unshifted subsignal is attenuated relative to the frequency-shifted subsignal, which is preferably amplified, before both subsignals are combined as part of the acoustic output. Considering both frequency bands as a pair, the likelihood of feedback is significantly reduced or eliminated.

Abstract: A hearing apparatus has a microphone for converting ambient sounds into a microphone signal, a signal processor for processing the microphone signal and an output transducer for outputting an output signal. A characteristic voice measure is ascertained for a voice component in the ambient sounds, and a supplementary measure characteristic of an activity of the wearer of the hearing apparatus is ascertained. Subsequently, an evaluated correlation between the voice measure and the supplementary measure is used to increase a probability value for the presence of a communication situation between the user and a third party if the voice measure and the supplementary measure, under at least one prescribed criterion, assume a value representative of the presence of the voice component and of the activity of the wearer. The probability value forms the basis for altering a signal processing algorithm that is executed to process the microphone signal.

Abstract: A method for the frequency distortion of an audio signal includes splitting the audio signal into a plurality of specified frequency bands, in which a band limit frequency is defined in each case by two respective immediately adjacent frequency bands. A first frequency band and a second frequency band lying immediately above the first frequency band are determined on the basis of the audio signal. A distortion of the frequencies differing from the distortion applied to signal components in the second frequency band is applied to signal components in the first frequency band, and a frequency-distorted signal is generated as a result. A method for suppressing an acoustic feedback in an acoustic system and a hearing aid are also provided.

Abstract: A method for reporting an aperiodic channel condition in a wireless communication system, according to an embodiment of the present invention, is performed by a terminal. The method comprises the steps of: receiving, from a base station, setting for one or more channel state information (CSI) processes including a plurality of channel state information-reference signals (CSI-RS) resources, wherein precoding is applied to each of the plurality of CSI-RS resources; receiving an aperiodic CSI report request from the base station; and transmitting, to the base station, an aperiodic CSI for a CSI process indicated by the aperiodic CSI report request, wherein, if a certain time has not elapsed after a CSI-RS resource index for the CSI process indicated by the aperiodic CSI report request is reported, the transmitted aperiodic CSI may include a CSI-RS resource index for a CSI process which has not been updated.

Abstract: A method and an apparatus reduce feedback in a hearing aid device. The method includes the step of acquiring a first feedback transfer function at a first point in time on a feedback path from a signal processing device via an electro-acoustic transducer, an acoustic signal path from the electro-acoustic transducer to an acousto-electric transducer and via the acousto-electric transducer back to the signal processing device. In a further step, a weighted mean value function is determined in a manner dependent on amplitude absolute values of the first feedback transfer function. A second feedback transfer function is estimated by an adaptive filter, wherein coefficients of the adaptive filter are determined in a manner dependent on the weighted mean value function. The adaptive filter is applied to a signal which is derived from an acoustic input signal of the acousto-electric transducer.

Abstract: A method for operating an electro-acoustic system (11) arranges an electro-acoustic device (10), for occluding an ear canal on an ear and uses a signal processing device (16) for processing a signal incoming at the device (10). A correction unit (17) of the signal processing device (16) modifies the signal incoming at the device (10). To reduce, to avoid or to compensate for an interfering or undesired change in a perception of ambient noises during the use of an electro-acoustic device occluding the ear canal, with the correction unit (17), a signal outgoing from the device (10) is generated in order to achieve acoustic transparency, in which, on the basis of the outgoing signal, a received signal is generated at the eardrum which is adapted so as to correspond to a free-ear received signal at the eardrum in the case of a free ear canal without the device (10).

Abstract: A method including operating an adaptive system of a hearing prosthesis, and determining one or more feedback path parameters of the hearing prosthesis based on the operation of the adaptive system of the hearing prosthesis, wherein the action of determining one or more feedback path parameters includes determining the one or more feedback path parameters based on data based on adaptive filter coefficients of adaptive filters of the adaptive system.

Abstract: A method of modelling a feedback path from a receiver to a microphone in a hearing device, includes: transmitting an electronic probe signal to the receiver; obtaining a microphone output signal that is associated with an output of the receiver; and determining at least one parameter of the feedback path based on the microphone output signal; wherein the electronic probe signal comprises a first signal level, followed by a second signal level that is less than the first signal level, the second signal level being greater than zero.

Abstract: Audio systems and methods are provided that detect instability in active feedback noise reduction circuitry. An acoustic transducer converts a driver signal into an acoustic signal, and a microphone provides a feedback signal. The feedback signal is processed, through a first transfer function, to provide an anti-noise signal. The driver signal is based at least in part upon the anti-noise signal, to reduce acoustic noise in the environment of the acoustic transducer. The driver signal is also filtered by a filter having a second transfer function that is inverse of the first transfer function, to provide a reference signal. The feedback signal is compared to the reference signal to determine a feedback instability, based upon the comparison.

Abstract: Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using and pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path.

Abstract: A method including obtaining data based on a current and/or anticipated future state of a hearing prosthesis and adjusting a set gain margin of the hearing prosthesis based on the current or anticipated future state of the hearing prosthesis.

Abstract: Disclosed herein, among other things, are apparatus and methods for annoyance perception and modeling for hearing-impaired listeners. One aspect of the present subject matter includes a method for improving noise cancellation for a wearer of a hearing assistance device having an adaptive filter. In various embodiments, the method includes calculating an annoyance measure or other perceptual measure based on a residual signal in an ear of the wearer, the wearer's hearing loss, and the wearer's preference. A spectral weighting function is estimated based on a ratio of the annoyance measure or other perceptual measure and spectral energy. The spectral weighting function is incorporated into a cost function for an update of the adaptive filter. The method includes minimizing the annoyance or other perceptual measure based cost function to achieve perceptually motivated adaptive noise cancellation, in various embodiments.

Abstract: A voice signal processing device connected to a smart device is provided. The voice signal processing device is connected to a smart phone, a tablet computer or the like and includes a voice signal input and output function mounted thereon. The voice signal processing device returns a sound generated by the smart device or a sound generated from the exterior into the smart device as a high-quality sound and is configured to extract a voice signal from a wire coupled to the smart device and the ear receiver and configured to input the extracted voice signal into the input terminal of the microphone of the smart device. A voice signal output from the smart device is configured to be extracted and conveniently input into the smart device and loss and distortion is minimized in the process of extracting and returning the voice signal.

Abstract: An in-ear audio device has a casing on which is disposed one or more bags are positioned to be filled with fillable material during a customization process in which a test sound is acoustically output into an ear canal by an acoustic driver of the in-ear audio device while the one or more bags are being filled, and a microphone acoustically coupled to the ear canal is employed to detect sounds within the ear canal that are indicative of the frequency response of the acoustic output of the acoustic driver to determine when the degree of sealing of the ear canal by the one or more bags is sufficient to achieve a desired quality of frequency response.