Abstract: In accordance with embodiments of the present disclosure, a method may include using an adaptive filter system to estimate a response of an electrical characteristic of a loudspeaker based on an error between a first electrical parameter of the loudspeaker and a second electrical parameter of the loudspeaker and adding a non-zero delay to the first electrical parameter relative to the second electrical parameter prior to calculation of the error such that the adaptive filter system captures a truncated non-causality of the electrical characteristic.

Abstract: A filter generator (100) generates a filter on the basis of band information (frequency) and gain characteristics (gain value) set by a user. The filter generator (100) obtains weighting factor information on the basis of the band information selected by the user and calculates a gain difference between a gain value used in a preceding filtering process and the new gain value selected by the user. The filter generator (100) then obtains a correction gain by multiplying the weighting factor information by the gain difference and generates a filter by multiplying a coefficient of the filter used in the preceding filtering process by the correction gain.

Abstract: A system and method are provided for controlling the own-voice experience of a user who talks while an earpiece is mounted to occlude an ear of the user. A vibration sensor is engaged with the user's head to sense vibrations formed by the user speaking and conducted by the user's head to the vibration sensor. A signal generator operates a speaker element, which is integrated with the earpiece, to generate sound waves in the ear canal of the user, based on the output signal of the vibration sensor. The signal generator generates the sound waves to at least partially cancel other sound waves that are generated by at least part of said vibrations entering the ear canal from surrounding bone and/or tissue. The system and method counteracts the occlusion effect and are applicable in connection with in-ear headsets, earphones, in-ear headphones, in-ear monitors, circumaural headphones, hearing aids, earplugs and earmuffs.

Abstract: Disclosed is an audio signal processing device comprising an input for receiving a noisy audio signal, a variable gain component and a noise suppression component. The noisy audio signal has a desired audio component and a noise component. The variable gain component and the noise suppression component are respectively configured to apply a gain and a noise suppression procedure to the audio signal, thereby generating a gain adjusted noise reduced audio signal. The aggressiveness of the noise suppression procedure is rapidly changed responsive to a change in the applied gain. That change is a change from a current value by an amount substantially matching the change in applied gain to a new value. The aggressiveness is then gradually returned to the current value.

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: The utility model provides a compressor system with an equalizer (EQ). The compressor system includes a signal input device in signal communication with the EQ and a compressor. The compressor system further includes a signal output device in signal communication with the compressor. The compressor includes a signal detection module. In the system, an electrical level threshold of a full-band audio signal is setup in the compressor, and the value of electrical level of the input signal is detected by the signal detection module. After comparison of electrical level threshold and the value of electrical level, the system determines whether the compressor is activated or deactivated. The system can achieve the technology effect of effectively compressing and limiting the audio signal of a different frequency.

Abstract: A hearing device, e.g. a hearing aid, is provided, comprising a forward path comprising an input transducer for providing an electric input signal, a signal processing unit configured to apply a requested forward gain to the electric input signal, and an output transducer. The hearing device further comprises a feedback reduction unit for reducing a risk of howl due to feedback from the output transducer to the input transducer. The forward path and the external feedback path defines a roundtrip loop delay. The feedback reduction unit is configured to modulate said requested forward gain in time, to provide that the resulting forward gain exhibits a first, increased gain AH in a first time period TH and a second, reduced gain AL in a second time period TL, wherein at least one of AH, AL, TH, and TL is/are determined according to a predetermined or adaptively determined criterion including said roundtrip loop delay.

Abstract: A hearing aid having hearing loss-compensating components, active occlusion reduction components, a vent, a tuned piston, and a flexible surround. The piston and the surround combination are assembled on the faceplate and cover the outside end of the vent that is situated on the faceplate. The piston and the surround combination minimize the adverse effects of walk-induced head vibrations.

Abstract: A far end signal from a far end user and broadcasting the far end signal at a loudspeaker is received. A signal for use in an echo transfer function is determined, and the signal selected from between: the far end signal or an echo reference signal received from a near field microphone disposed in close proximity to the speaker. The near field microphone sensing the far end signal that is broadcast from the speaker while sensing the near-end speech and ambient noise at insignificant energy levels compared to the speaker signal. An echo transfer function is determined based at least in part upon the selected signal, and the echo transfer function represents characteristics of an acoustic path between the loudspeaker and a far field microphone that is disposed at a greater distance from the speaker than the near field microphone. An estimated echo is determined based at least in part upon the echo transfer function.

Abstract: A system and method for selectively enhancing an audio signal to make sounds, particularly speech sounds, more distinguishable. The system and method are designed to divide an input auditory signal into a plurality of spectral channels having associated unenhanced signals and perform enhancement processing on a first subset of the spectral channels and not perform enhancement processing on a second subset of the spectral channels. The enhancement processing is performed by determining an output gain for at least the first subset of spectral channels based on a time-varying history of energy of the unenhanced signals associated with each channel in the first subset of the spectral channels and applying the output gain for each of the first subset of the spectral channels to the unenhanced signals to form enhanced signals associated with each of the first subset of the spectral channels.

Abstract: Disclosed herein, among other things, are methods and apparatus for allocating feedback cancellation resources for improved acoustic feedback cancellation for hearing assistance devices. In various embodiments, a hearing assistance device includes a microphone and a processor configured to receive signals from the microphone and process them according to a plurality of processing blocks. The processor is adapted to include an event detector that can provide detection of an event and an output to adjust one or more processing blocks of the overall process to more efficiently use resources of the processor for the event detected, in various embodiments.

Abstract: The present application discloses systems and methods to analyze feedback path information during a fitting session. In accordance with one embodiment, a method is provided and includes during a fitting session, calculating a feedback gain margin of a hearing prosthesis by causing the hearing prosthesis to receive a test signal, output an output signal based on the test signal, and receive a feedback signal based on the output of the output signal, the test signal being configured to test a different parameter of the hearing prosthesis.

Abstract: A new method for performing adaptive feedback suppression in a hearing aid and a hearing aid utilizing the method are provided. According to the method, a slow adaptive filter and a fast adaptive filter with different error signals for filter coefficient updating are used for feedback suppression.

Abstract: A system for suppressing signal harmonic distortion caused by a main signal processing device provides processing for an electrical signal containing sine components of multiple frequencies. A post-processor reduces the amplitude of each frequency, relative to other frequencies, as the frequency increases, thus suppressing harmonics, which have higher frequencies than their respective elementary signals. To compensate for frequency distortion caused by the differential frequency suppression in the post-processing stage, a pre-processor provided upstream of the main signal processing device provides frequency-dependent signal processing having a generally opposite effect to that of the post-processor, increasing the amplitude of each frequency of the signal passing through the pre-processor, relative to other frequencies, as the frequency increases. The resulting output signal thus has substantially the same frequency spectrum as the original signal while distortion-causing harmonics are reduced or eliminated.

Abstract: An apparatus for determining cochlear dead region comprising a sound generation unit, an output transducer, and a control unit. The control unit causes the sound generation unit to provide a first electrical sound signal representing a first frequency band adjacent to a frequency region of interest, with a first bandwidth, a first sound pressure level and a first subsequent electrical sound signal representing a first subsequent frequency band comprising the frequency region of interest with a first subsequent bandwidth, a first subsequent sound pressure level, and a first subsequent time delay to the first electrical sound signal. In response, the output transducer generates a first output sound and a first subsequent output sound with the first subsequent time delay. A user compares the first output and first subsequent output sounds and decides, whether he hears a difference between the two sounds and gives a positive or negative decision input.

Abstract: A method may include forming a plurality of beams to receive and spatially filter audible sounds from a first microphone and a second microphone. The plurality of beams may include a first unidirectional beam to receive audible sounds from the first microphone and the second microphone, a second unidirectional beam to receive audible sounds from the first microphone and the second microphone, the second unidirectional beam having a spatial null in a direction different from that of the first unidirectional beam, and an omnidirectional beam to receive audible sounds from the first microphone and the second microphone. The method may also include determining a first energy of audible sounds filtered by the first unidirectional beam and a second energy of audible sounds filtered by the second unidirectional beam and, based on the first energy and the second energy, selecting one of the plurality of beams as a selected beam.

Abstract: Moving armature receiver assemblies wherein a first U-shaped armature and a second U-shaped armature are configured for suppression of vibration of a housing structure along a longitudinal housing plane. The first and second U-shaped armatures may be shifted away from each other along a longitudinal housing plane to render the first and second U-shaped armatures partially overlapping in the orthogonal plane with a predetermined overlap distance.

Abstract: Feedback whistling and external tonal signals are distinguished during feedback suppression. For that purpose an adaptation increment of an adaptive filter of a hearing apparatus for feedback reduction is controlled. A sound signal is picked up by a microphone and a microphone signal is output, from which an earpiece signal for an earpiece is generated. An adaptive filter reduces a feedback signal in the microphone signal. To this end an autocorrelation value of sampled values of the microphone signal, between which a time difference exists, is obtained, and the adaptation increment of the adaptive filter is controlled based on the autocorrelation value. A frequency of an output signal obtained on the basis of the microphone signal is shifted while creating the earpiece signal and the time difference for obtaining the autocorrelation value is controlled as a function of the shifting of the frequency of the microphone signal.

Abstract: A hearing aid according to an embodiment of the present disclosure includes: a microphone configured to convert sound into an electric signal and output a first signal; a receiver configured to convert the third signal that is generated by performing predetermined hearing aid processing on a second signal, into sound; a first feedback removal unit having an adaptive filter configured to adaptively estimate a first transfer function of a sound feedback path from the receiver to the microphone; and a second feedback removal unit having a fixed filter configured to receive the third signal and use a fixed filter coefficient based on a second transfer function of a vibration feedback path from the receiver to the microphone. The second signal is generated by removing from the first signal, output signals from the adaptive filter and the fixed filter.

Abstract: Acoustic feedback in an audio device is reduced or eliminated via a feedback compensator. The feedback compensator disposes of at least one adaptive filter whose adaptation is governed by a step-size parameter ?. A step-size parameter ? of the adaptive filter is set at a high level to eliminate acoustic feedback and the parameter ? is decreased when a critical excitation signal is detected. This way to proceed allows for efficient elimination of feedback whistling while required environment signals are retained.

Abstract: An elastic body of an audio accessory is provided. The elastic body includes a cylindrical inner cover, at least one outer cover which is elongated from a top end edge of the cylindrical inner cover to enclose the cylindrical inner cover, the at least one outer cover having at least one inlet which partially exposes at least a part of the cylindrical inner cover, and at least one sensor module disposed on at least one of the cylindrical inner cover and the at least one outer cover and configured to generate a sensor signal related to deformation of the at least one outer cover.

Abstract: A hearing aid device is disclosed. The hearing aid device comprises a receiver configured to be arranged in the ear canal. The hearing aid device comprises a directional microphone system comprising two microphones or one microphone having two sound inlets. The hearing aid device comprises means for counteracting acoustic feedback on the basis of sound signals detected by the two microphones or the two sound inlets. The hearing aid device comprises an “open fitting” providing ventilation. The two microphones or the two sound inlets of the directional microphone (forming part of a directional system) are arranged in the ear canal at the same side of the receiver and sound is allowed to propagate freely between the microphones or between the inlets of the directional microphone and the receiver. Preferably, the hearing aid device comprises a procedure (such as an adaptive procedure) for optimizing the directional system of the hearing aid device.

Abstract: An audio signal processing system and an echo signal removing method thereof are provided. The audio signal processing system includes a speaker that is configured to output an audio signal; a microphone that is configured to receive the audio signal output by the speaker including an echo signal generated by the audio signal; an echo signal delay unit that is configured to delay the echo signal for a bulk delay time, and output the echo signal that is delayed; and an echo signal removing unit that is configured to remove the echo signal that is delayed and output by the echo signal delay unit from the audio signal received by the microphone, wherein the echo signal delay unit includes a bulk delay measuring unit that is configured to measure a bulk delay by analyzing impulse response characteristics of an echo path.

Abstract: Monitoring usage of a hearing device includes: c) checking whether a message has been received within a time period (dt) shorter than a pre-defined time period (Dt) via a long-distance communication network in a remote device; and d) generating usage status data dependent on the result of the checking addressed in step c). The usage status data are indicative of the usage of the hearing device, indicating “use” if the time period (dt) is smaller than or equal to the pre-defined time period (Dt) and indicating “non-use” if the time period (dt) is larger than the pre-defined time period (Dt). The message includes identification data for identifying the hearing device or a user of said hearing device. The remote device is located remote from the hearing device and connected to the long-distance communication network. The time period (dt) starts from a start event.

Abstract: Methods and apparatus for fitting a hearing aid device (3) that includes a part which is arranged in the ear canal (2) of a user (31).

Abstract: When hearing-aid processing and predetermined-sound suppression processing are performed, in order to reduce an increase in delay of an output sound in response to an input sound greater than a case where only the hearing-aid processing is performed, a hearing aid includes: a hearing-aid processing unit configured to amplify a first acoustic signal received by a microphone, according to an ability of a user to hear sound in each of frequencies, to produce a second acoustic signal; a predetermined-sound detection unit configured to detect a predetermined sound included in the first acoustic signal, and to produce a control signal indicating generation timing of the predetermined sound; and a predetermined-sound suppression unit configured to suppress the second acoustic signal produced in the hearing-aid processing unit at the generation timing of the predetermined-sound indicated by the control signal produced in the predetermined-sound detection unit.

Abstract: Provided are a method and apparatus for reproducing sound. A sound signal may be generated by receiving sound around the apparatus. An output signal may be generated by performing a process according to functions of the apparatus on a residual signal obtained by canceling a feedback signal from the sound signal.

Abstract: A hearing apparatus and method for measuring a distance between an eardrum and the hearing apparatus are provided. The hearing apparatus may include a signal output unit configured to output a first measurement signal, a signal identification unit configured to receive a second measurement signal, and a distance determination unit configured to determine the distance between the user's eardrum and the hearing apparatus.

Abstract: This invention concerns a method, and a device, for feedback cancellation. This invention also concerns a computer program product comprising computer program code means to make a computer execute a procedure for feedback cancellation. The method comprises providing an adaptive feedback cancellation filter which adapts under the control of a control module, and filtering at least one input of the control module to suppress correlated signals from the input prior to the control module operating upon the input. The device comprises an adaptive feedback cancellation filter, a control module and at least one filter. The control module controls adaptation of the adaptive feedback cancellation filter. The filter suppresses correlated signals from an input to the control module prior to the control module operating upon the input.

Abstract: A headset having a talk-through microphones incorporates an audio circuit that compresses a signal representing sounds detected by the talk-through microphones in response to the audio circuit detecting the onset of a peak (positive and/or negative) in the signal that exceeds a predetermined voltage level (positive and/or negative voltage level, perhaps a predetermined magnitude of voltage from a zero voltage level), and that does so with a rate of change in voltage level that exceeds a predetermined rate of change in voltage level, the degree of compression possibly being a compression to or near a zero amplitude (perhaps to or near a zero voltage level) and the duration of the compression possibly being controlled by a timing circuit set to a predetermined period of time that may be retriggerable while amidst the predetermined period of time.

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 method provides a long term feedback path estimate of a listening device. The method comprises a) providing an estimate of the current feedback path; b) providing a number of detectors of parameters or properties of the acoustic environment of the listening device and/or of a signal of the listening device, each detector providing one or more detector signals; c) providing a criterion for deciding whether an estimate of the current feedback path is reliable based on said detector signals; d) storing said estimate of the current feedback path, if said criterion IS fulfilled and neglecting said estimate of the current feedback path, if said criterion is NOT fulfilled; e) providing a long term estimate of the feedback path based on said stored estimate(s) of the current feedback path.

Abstract: A hearing device comprises a feedback-path estimation unit, which adaptively estimates a feedback path from an output transducer to an input transducer, and sets an adaptation-step size of an adaptive feedback-path estimation algorithm in dependence on an estimate of a background-noise spectrum. The feedback-path estimation unit provides an estimation-control signal for generating an acoustic feedback-path estimation signal having an level spectrum, which has at least one first frequency band with non-zero level and at least one second frequency band with zero level or with a non-zero level smaller than the level in the first frequency band and smaller than a background-noise level in the respective second frequency band. Background noise is detected in the second frequency band while the feedback-path estimation signal is provided. The background-noise level in the at least one first frequency band is estimated for obtaining the estimate of the background-noise spectrum. A corresponding method is described.

Abstract: FM and AM receivers in car environments require a noise blanker circuit or method to reduce the audible disturbances generated by impulse noise, such as ignition noise, in the audio signal. The invention proposes a combination of time-domain and frequency-domain processing to reduce the audible distortion. The time-domain processing interpolates the signal during the impulse noise bursts, and the frequency-domain processing disperses the remaining signal distortion over time, or relocates it to positions in time where it is attenuated by a windowing technique.

Abstract: Disclosed is an ear microphone which includes: a microphone for converting a voice signal provided from the external auditory meatus of a user to an electric signal; a speaker for converting the electric signal provided from an external device to the voice signal; and a soundproof member having a microphone receiving groove for fixing and supporting the microphone to the inside of the housing and a speaker receiving groove for fixing and supporting the speaker to the inside of the housing.

Abstract: A listening device includes a) an input transducer; and b) an output transducer for converting a processed electric signal to an output sound; a forward signal path being defined there between and comprising c) a signal processing unit for processing an electric input signal or a signal derived therefrom and providing a processed output signal; d) a manually operable user interface located on the listening device allowing a user to control a function of the listening device; e) a feedback estimation system for estimating a feedback path from the output transducer to the input transducer, the feedback estimation system comprising e1) an adaptive filter having e11) a variable filter part and e12) an algorithm part comprising an adaptive algorithm.

Abstract: A hearing aid, signal processing method and program enables TV sound to be made easier for a hearing aid user to hear when wishing to watch TV, and a person's voice to be made easier to hear when wishing to converse with that person. The hearing aid includes another-person's speech detection section that detects speech of a speaker other than the wearer using detected sound source direction information, an own-speech detection result, and a TV sound detection result, and a per-sound-source frequency calculation section that calculates the frequency of each sound source using an own-speech detection result, TV sound detection result, other-speaker's speech detection result, and sound source direction information. A scene determination section determines a scene using sound source direction information and a per-sound-source frequency; and an output sound control section controls hearing of the hearing aid according to a determined scene.

Abstract: Provided is a hearing aid system including: a first processor that fast Fourier transforms N input signal tone data output from an input buffer memory, and then executes nonlinear compression; a second processor that inverse fast Fourier transforms amplitude spectrum data; an output buffer memory that stores the voice signal tone data, until the number of the voice signal tone data is N; and a digital-to-analog (D/A) converter that converts the digital voice signal tone data into an analog signal, to then output the analog signal to a receiver. Thus, certain ambient noise due to an acoustic feedback signal and a narrow frequency band that occur in a hearing aid is removed, to thus reduce discomforts due to the acoustic feedback noise of the hearing aid for hearing aid users, and to thereby significantly improve speech discrimination.

Abstract: A hearing device comprises a feedback-path estimation unit, which adaptively estimates a feedback path from an output transducer to an input transducer, and sets an adaptation-step size of an adaptive feedback-path estimation algorithm in dependence on an estimate of a background-noise spectrum. The feedback-path estimation unit provides an estimation-control signal for generating an acoustic feedback-path estimation signal having an level spectrum, which has at least one first frequency band with non-zero level and at least one second frequency band with zero level or with a non-zero level smaller than the level in the first frequency band and smaller than a background-noise level in the respective second frequency band. Background noise is detected in the second frequency band while the feedback-path estimation signal is provided. The background-noise level in the at least one first frequency band is estimated for obtaining the estimate of the background-noise spectrum. A corresponding method is described.

Abstract: A hearing aid includes: an input transducer for generating an audio signal; a feedback suppression circuit configured for modelling a feedback path of the hearing aid; a subtractor for subtracting an output signal of the feedback suppression circuit from the audio signal to form a feedback compensated audio signal; a signal processor that is coupled to an output of the subtractor for processing the feedback compensated audio signal to perform hearing loss compensation; and a receiver that is coupled to an output of the signal processor for converting the processed feedback compensated audio signal into a sound signal; wherein the hearing aid further comprises a gain processor for performing gain adjustment of the feedback compensated audio signal based at least on an estimate of a residual feedback signal of the feedback compensated audio signal, wherein the estimate of the residual feedback signal is based at least on the audio signal.

Abstract: Disclosed are methods and systems for informing a user about hearing aid feedback noise including, for example, receiving, through one or more microphones, an acoustic signal sample, analyzing the acoustic signal sample to determine whether feedback noise is present in the acoustic signal sample, and displaying to the user an indication of whether feedback noise is present in the acoustic signal sample.

Abstract: Systems, methods, and devices are disclosed. An example sound processor is disclosed. The example sound processor includes a module configured to identify a feedback artifact in a current sample of an input spectral component by determining that a change in a signal level of an input spectral component is approximately equal to a predicted change. The predicted change may be based on one or more characteristics of an external feedback loop. Responsive to identifying the feedback artifact, the sound processor is further configured to apply a modification to a parameter used to process the input spectral component. The modification reduces a likelihood of including audible feedback in a processed input spectral component.

Abstract: An apparatus related to earmolds with venting configurations designed to relieve the occlusion effect. Various designs provide multiple vents allow residual ear canal air volume to vent to and from air outside the ear and the earmold. In various designs, the earmold includes one vent between the residual ear canal air volume and a volume of air internal to the earmold. A second vent provides passage of air internal to the earmold and air external to the ear and the inserted earmold when worn by a user.

Abstract: A method for controlling adaptation of a feedback suppression filter in a hearing aid comprises the step of suspending the adaptation of the feedback suppression filter when the level of a hearing aid reference signal is below a predetermined threshold. The invention further provides a hearing aid with a feedback suppression filter and with means for suspending adaptation of the feedback suppression filter.

Abstract: Provided is an electronic device including: a housing; a vibration body disposed in the housing; and a microphone disposed in the housing. By vibrating the vibration body, the electronic device transmits air conduction sound and vibration sound that is generated by the vibration. During the generation of the sounds, howling reduction processing is performed, so that usability of the electronic device is improved.

Abstract: A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.

Abstract: The application relates to a hearing assistance device (HAD) comprising (a) an input transducer system comprising (a1) an audio input transducer (AIT), and (a2) a first supplementary input transducer (SIT1), (b) an output transducer (OT) for converting a processed output signal to a stimulus perceivable by said user as sound, and (c) a signal processing unit (SPU) operationally connected to said audio input transducer (AIT), to said first supplementary input transducer (SIT1), and to said output transducer (OT), said signal processing unit (SPU) being configured for processing said electric audio input signal, and said first supplementary electric input signal, and for providing said processed output signal. The audio input transducer (AIT) is adapted for being located in an ear of the user. In a NORMAL mode of operation, electric audio input signal is processed in the signal processing unit and the supplementary electric input signal(s) are used to control the processing.

Abstract: A hearing aid includes: a microphone; a hearing aid processing unit configured to provide a gain to a first signal based on an output signal from the microphone to generate a second signal; a receiver configured to convert the second signal into sound; an adaptive filter configured to adaptively estimate a transfer function corresponding to a pathway from an input side of the receiver to an output side of the microphone; and a feedback removal unit configured to subtract a third signal generated based on the transfer function from the output signal of the microphone to obtain a signal and output the signal as the first signal; and a control unit configured to control the gain setting unit and an adaptive speed of the adaptive filter.

Abstract: The application relates to a hearing assistance device and to a method of performing a real ear measurement. The method comprises providing a first controlled acoustic feedback path from an output transducer to a measurement input transducer via a standard acoustic coupler; generating a first probe signal; estimating and storing a first estimate of the first controlled acoustic feedback path; and providing a second controlled acoustic feedback path from the output transducer to the measurement input transducer via the residual volume between the ITE part of the hearing aid device and the user's eardrum; generating a second probe signal; estimating and storing a second estimate of the second controlled acoustic feedback path; and determining a real ear to coupler difference from said first and second acoustic feedback estimates. An alternative and relatively simple method of determining an RECD-value in hearing assistance device of a particular user is thereby provided.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.