Abstract: High-quality adjustment of a hearing apparatus, and in particular of a hearing aid, is accelerated. In the adjusting method, a percentile analysis is carried out in order to obtain at least one frequency response of the hearing apparatus. Any discrepancy between the frequency response and an intended gain characteristic is determined. The setting of the hearing apparatus is then changed, such that the discrepancy is reduced. In order to check the effect of the setting change, a current frequency response after variation of the setting is determined by providing a constant noise signal or a single sinusoidal sweep at the input of the hearing apparatus and level measurement at the output. This frequency response measurement can be carried out as a second step considerably more quickly than the very precise percentile analysis at the start of the process.

Abstract: Artifacts are reduced during a frequency compression of an audio signal in a hearing device, in particular in a hearing aid. An amplitude information of a source channel is obtained, for example a spectral envelope, from several frequency channels of the audio signal. An amplitude corresponding to the amplitude information is then applied on a signal in a target channel of the several frequency channels, on which the source channel is represented during frequency compression.

Abstract: A hearing aid device suitable for executing a frequency compression is optimized to achieve a better assignment between channels of a source frequency range and channels of a target frequency range. A frequency range that can be transmitted by the hearing aid device is split into a number of frequency bands, after which a number of adjacent frequency bands are combined into at least one group of frequency bands and then one frequency band is selected from each group of frequency bands. The selected frequency bands are moved respectively into a target frequency band permanently assigned to the respective group. This prevents that a number of frequency bands from the source frequency range are shifted to the same target frequency band.

Abstract: A disclosed method selects a plurality of fewer than all of the channels of a multichannel signal, based on information relating to the direction of arrival of at least one frequency component of the multichannel signal.

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 can be tuned while worn by a user according to a heuristic method, including select an adjustable parameter and an stimulus having audio energy characteristics related to the adjustable parameter. The stimulus is played via a calibrated audio output device and amplified by the hearing aid. After selecting a user perception query associated with the adjustable parameter, feedback is obtained from the user indicating that the stimulus was not perceived neutrally. A perceptual adjustment curve and a threshold curve are determined according to the user feedback, and an estimated perception curve that estimates the user's perception of the stimulus is determined. When at least a portion of the estimated perception curve exceeds a threshold curve, the hearing aid is programmed according to at least one adjustment value determined according to the user feedback value.

Abstract: A method is provided for adapting a feedback suppression device of a hearing device to a given situation in order to improve the quality of feedback suppression in hearing devices and in hearing aids in particular. In the method an adaptation procedure of the feedback suppression device is periodically activated and adaptation of the feedback suppression device is performed regularly even if a feedback detector does not detect a feedback situation. A feedback suppression device for a hearing device is also provided.

Abstract: With a hearing aid device suitable for executing a frequency compression, the aim is for different input signals entering the hearing aid device to be easily distinguishable even after frequency compression. To this end it is proposed that form factors should be determined from source frequency bands in which the signal components of the input signal present therein are not included in the output signal, these being included in the amplification of selected frequency bands, the signal components of which are included in the output signal.

Abstract: A method for the operation of a hearing device with at least one receiver includes carrying out a fixed second frequency shift of an electrical receiver signal supplying the receiver or of a predefinable frequency range of the receiver signal and a variable first frequency shift of the receiver signal or of a predefinable frequency range of the receiver signal. The first frequency shift is changed depending on the occurring feedback. A frequency shift which effectively prevents feedback is advantageously carried out, while artifacts of the frequency shift are minimized.

Abstract: The present application is directed to a hearing aid apparatus for wearing use by a user, including a frontend sound collector configured to collect a frontend signal; a backend sound collector configured to collect a backend signal; and a sound processor configured to process the frontend signal and the backend signal; wherein the sound processor includes a frontend delayer configured to apply a delay coefficient to the frontend signal to produce a delayed frontend signal; a backend delayer configured to apply the delay coefficient to the backend signal to produce a delayed backend signal; and an adaptive filter configured to process the delayed frontend signal and the delayed backend signal to produce an adaptive filter output signal.

Abstract: A method for enhancing speech output in real time is used in a hearing aid device. The input speech is divided into multiple audio segments first. Then each audio segment is analyzed for its attribute: high frequency, low frequency, or soundless. Low frequency segments are outputted without undergoing frequency processing. High frequency segments are outputted after undergoing frequency processing. All or some of the soundless segments are deleted without being outputted. The deletion of soundless segments can reduce the delay caused by the frequency processing of the high frequency segments.

Abstract: A method of processing a signal in a hearing aid (50) comprising the steps of determining a gradient of a speech intelligibility measure using a closed form expression of the gradient and modifying the frequency dependent hearing aid gains in dependence on the calculated gradient. The invention also provides a hearing aid (50).

Abstract: An FSK transceiver for use in a hearing aid has means for automatic tuning of the transceiver frequency. The automatic tuning means comprises a phase difference detector and a tuning manager adapted to control the frequency of a resonant circuit of the FSK transceiver according to information received from the phase difference detector. The phase difference detector measures the phase difference across a component of the resonant circuit of the FSK transceiver. If the phase difference of a binary FSK symbol is equal to the phase difference of the opposite binary FSK symbol, the resonant circuit is tuned, otherwise the tuning manager performs a tuning procedure in order to minimize the difference in phase differences, and adjusts the frequency of the resonant circuit accordingly.

Abstract: Disclosed are an apparatus and a method for transmitting human sound that cancel an interference signal generated by a non-linear property of a human body around an ear by synthesizing and transmitting a sound signal modulated by a single side band (SSB) transmission method and a carrier to one output unit and by synthesizing and transmitting the sound signal modulated by the single side band (SSB) transmission method and a carrier in which a phase of a signal is transited to the other output unit, by using two signal output units in a sound transmitting apparatus.

Abstract: Methods, systems, and devices for processing an audio signal are provided. An example method includes mapping a fundamental frequency of an audio signal to a modulation frequency. An output of the mapping is less than the fundamental frequency when the fundamental frequency is greater than an intersection frequency. The intersection frequency is a frequency at which the output of the mapping is the fundamental frequency.

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: The present invention is directed to a method and a hearing device for extending a usable frequency range of an analog input signal (i) being processed by a hearing device, the method comprising the steps of converting the analog input signal (i) to a first output signal (o1) and to an intermediate signal (om), the first output signal (o1) having a final sampling rate and the intermediate signal (om) having an intermediate sampling rate that is greater than the final sampling rate, applying a band-pass filter unit (31) to the intermediate signal (om) in order to obtain a filtered intermediate signal (omf), a lower cut-off frequency of the band-pass filter unit (31) being above half the final sampling rate, an upper cut-off frequency of the band-pass filter unit (31) being below half the intermediate sampling rate, and shifting a spectrum of the filtered intermediate signal (omf) to a frequency range being below the final sampling rate to obtain an intermediate output signal (om2).

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: An audio system includes a headset and a portable computing device. The headset includes a first speaker and a first microphone, a second speaker, a second microphone and a processing unit. The first microphone and the second microphone output a first signal and a second signal, respectively. The processing unit combines the first electrical signal and the second electrical signal into a third electrical signal, and communicates the third electrical signal to a portable computing device through an output channel. The portable computing device preferably includes an application that separates the third electrical signal into the first electrical signal and the second electrical signal. The application compensates the first electrical signal and the second electrical signal for a hearing loss of an individual.

Abstract: A hearing device includes a first filter configured for providing a first frequency part of an input signal of the hearing device, the first frequency part comprising a low pass filtered part, a second filter configured for providing a second frequency part of the input signal, the second frequency part comprising a high pass filtered part, a first synthesizing unit configured for generating a first synthetic signal from the first frequency part using a first model based on a first periodic function, and a combiner configured for combining the second frequency part with the first synthetic signal for provision of a combined signal.

Abstract: A method for adjusting a hearing system to the hearing preferences of a user of said hearing system is disclosed, wherein said hearing system is capable of carrying out frequency transposition of audio signals, which frequency transposition depends on at least one parameter. The method comprises the steps of A) carrying out a distinction test for examining said user's ability to distinguish between two stimulus signals which differ in their frequency contents; B) adjusting said at least one parameter in dependence of the result of said distinction test. The distinction test comprises the steps a2) consecutively playing said two stimulus signals to said user; b2) receiving from said user information indicative of whether said user perceived said two stimulus signals as two times the same sound or as two different sounds; c2) deriving a value from said information received from said user; wherein said result of said distinction test is dependent on said value.

Abstract: Disclosed herein, among other things, are apparatus and methods for a threshold-derived fitting rationale using frequency translation for hearing assistance devices. In various method embodiments, a first audiogram is received for a first hearing assistance device for a wearer, and a second audiogram is received for a second hearing assistance device for the wearer. The first audiogram and the second audiogram are compared to audiometric thresholds to determine if frequency translation should be enabled.

Abstract: An embodiment of a hearing assistance apparatus for performing a Real Ear Measurement (REM), comprises a hearing assistance device housing, a microphone within the housing, an earhook connected to the housing, and a flexible tube. The house has a first opening for guiding sound into the housing to the microphone. The housing and the connected earhook form an interface, where the earhook has a shape to provide a slot near the interface of the housing and the earhook. The tube guides sound, and has a first end and a second end. The first end of the flexible tube and the slot of the earhook cooperate to retain the first end of the flexible tube in the slot of the earhook and flush with the housing to provide a sound-tight connection with the first opening.

Abstract: A system and method for using an audiogram and audio filters to provide improved audio characteristics for hearing impaired listeners by equalizing the received sound level of the signal across the frequency spectrum. The system and method further provide for modifying an audio signal based on the personal hearing characteristics of a listener to compensate for hearing loss.

Abstract: A suppressed carrier audio system can include a modulator having an input configured to receive an audio signal having audio content and configured to modulate the received audio signal onto an ultrasonic carrier to produce a modulated signal; a bandpass filter to receive the modulated signal and suppress or remove the carrier from the modulated signal, and further configured to pass a sideband of the modulated signal thereby creating a suppressed carrier signal; and a first ultrasonic transducer having an input coupled to receive the suppressed carrier signal, the ultrasonic transducer configured to emit the suppressed carrier signal in a direction toward an intended listener. The system can also include a demodulator having a signal generator configured to generate a carrier signal and a second ultrasonic transducer having an input coupled to receive the carrier signal and to emit the carrier signal in a direction toward the intended listener.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: The understanding of speech is to be improved in cases of hearing loss in which certain frequency ranges can no longer be perceived even at high volume or amplification, as applicable. A frequency transposition is performed on an input signal coming into a hearing aid and specific features of an electrical input signal are detected and a transposed signal is filtered as a function of the specific features detected in the electrical input signal. The hearing aid provides appropriate facilities for detecting specific features of the electrical input signal and filtering facilities for filtering the transposed signal. The objective of this is to detect particular properties of the input signal, in particular signal properties which are characteristic of fricatives, which may possibly be lost as a result of the frequency transposition, and at least partially to restore them in the transposed signal.

Abstract: The invention relates to a method of operating an audio processing device. The invention further relates to an audio processing device, to a software program and to a medium having instructions stored thereon. The object of the present invention is to provide improvements in the processing of sounds in listening devices. The problem is solved by a method comprising a) receiving an electric input signal representing an audio signal; b) providing an event-control parameter indicative of changes related to the electric input signal and for controlling the processing of the electric input signal; c) storing a representation of the electric input signal or a part thereof; d) providing a processed electric output signal with a configurable delay based on the stored representation of the electric input signal or a part thereof and controlled by the event-control parameter. The invention may e.g. be used in hearing instruments, headphones or headsets or active ear plugs.

Abstract: A BTE hearing aid includes a BTE hearing aid housing, at least one BTE sound input transducer, a processor configured to generate a hearing loss compensated output signal, a sound signal transmission member for transmission of a signal from a sound output of the BTE hearing aid housing to an ear canal of a user at a second end of the sound signal transmission member, an earpiece configured to be inserted in the ear canal, an output transducer, and an ITE microphone housing accommodating at least one ITE microphone, wherein the ITE microphone housing is configured to be positioned in an outer ear, wherein the processor is further configured for processing an audio signal from the at least one ITE microphone and an audio signal from the at least one BTE sound input transducer in such a way that the hearing loss compensated output signal substantially preserves spatial cues.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: A binaural hearing apparatus has at least one left hearing device and at least one right hearing device. The apparatus further includes a first frequency distortion unit in the left hearing device, which distorts the frequency of an acoustic signal received by the left hearing device or a signal part of the received acoustic signal and a second frequency distortion unit in the right hearing device, which distorts the frequency of the acoustic signal received by the right hearing device or a signal part of the received acoustic signal, with the frequency distortions of the left and right hearing device being different. This is advantageous in that the subjective perception of superimposition artifacts by a hearing device wearer is reduced.

Abstract: Among other things, a sound processing device system is disclosed to assist a hearing-impaired human listener recognize speech sounds or phonemes. The device system may be configured at least to generate an output audio signal at least by transposing and causing a negative rank ordering of frequency of at least a portion of the input audio signal. Compression also may be performed on the at least the portion of the input audio signal as part of generating the output audio signal. The negative rank ordering may be performed on a high-frequency portion of the input audio signal that becomes a low-frequency portion of the output audio signal by the transposing. The low-frequency portion of the output audio signal may represent an inverted ordering of frequencies or frequency segments present in the high-frequency portion of the input audio signal.

Abstract: A hearing loss compensation apparatus includes a filter bank control unit configured control a plurality of filter banks to separate an input signal into frequency bands; and a signal output unit configured to amplify the input signal separated into frequency bands according to a control parameter based on characteristics of a hearing loss patient, and output the amplified input signal.

Abstract: A method and a hearing system for adaptively matching microphones (3, 4) of a hearing system. The method comprising the steps of determining a true direction towards a sound source, determining an estimated direction towards the sound source using microphones (3, 4) of the hearing system, comparing the true direction with the estimated direction to obtain a correction factor, applying the correction factor to the signals of the microphones (3, 4) of the hearing system in order to reduce a difference between the true direction and a corrected estimated direction obtained via corrected microphone signals.

Abstract: A method for enhancing the accuracy of sounds heard by a hearing-impaired listener is disclosed. The method for enhancing the accuracy of sounds heard by a hearing-impaired listener includes receiving an input sound, determining if it is necessary to modify the frequency of the input sound, and modifying the input sound into a modified input sound if necessary. The determination relies on the frequency and energy of the input sound. The ratio of the energy of lower frequencies of the modified input sound will be increased.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a computing device having a controller to modify sound produced by a gaming application according to an otological profile of a user. Additional embodiments are disclosed.

Abstract: Feedback whistle in hearing devices is intended to be able to be suppressed without loss of output of the useful signal. To this end, it is provided to establish or predetermine a frequency range which is susceptible to feedback. From an input signal which has a spectral component in the frequency range susceptible to feedback, a predeterminable component is substituted with a synthetic signal. Mixing-in a synthetic signal is also possibly used to widen the spectrum of an input signal, which is limited.

Abstract: In a digital hearing aid device (1) frequency modification is employed above a lower spectral bound and in accordance with a compression factor. The frequency modification is dynamically adjusted in dependence on a sound environment analysis (10) or an end-user input (30), by modifying the frequency modification parameters such as a lower spectral bound and a compression factor. The adjustment can be based on an interpolation between predefined parameters. In certain sound environments, such as loud noise, own-voice and telephone conversations, frequency modification is reduced or switched off. The proposed solutions have the advantage that the occurrence of disturbing noise and of distortions of harmonic relationships at the end-user's ear is reduced and signal processing resources as well as battery resources are saved.

Abstract: A hearing device has a feedback suppression unit. The hearing device further has a low-pass filter characterized by a first cut-off frequency, which couples out a low-frequency signal component from an output signal of the hearing device, and a high-pass filter characterized by a second cut-off frequency, which couples out a high-frequency signal component from the output signal of the hearing device. A frequency shift unit shifts the frequency of the high-frequency signal component to higher frequencies. A gap exists between the first and the second cut-off frequency. As a result of the different limit frequencies, signal distortions caused by frequency shifts are effectively suppressed. Feedback is suppressed continuously and rapidly at higher frequencies.

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: A listening device includes an ear-part for being worn in or at an ear of a user, a microphone system including at least two microphones each converting an input sound to an electrical microphone signal, and a TF-conversion unit for providing a time-frequency representation of the at least two microphone signals. Each signal representation includes complex or real values of the signal in a particular time-frequency unit. The listening device also includes a DIR-unit with a directionality system providing a weighted sum of the at least two electrical microphone signals thereby providing at least two directional microphone signals having maximum sensitivity in spatially different directions and a combined microphone signal. Each time-frequency unit of the combined signal is attributable to a particular direction. A frequency shaping-unit modifies one or more selected time-frequency units to indicate directional cues of input sounds providing an improved directional output signal.

Abstract: A BTE hearing aid includes a BTE hearing aid housing, at least one BTE sound input transducer, a processor configured to generate a hearing loss compensated output signal, a sound signal transmission member for transmission of a signal from a sound output of the BTE hearing aid housing to an ear canal of a user at a second end of the sound signal transmission member, an earpiece configured to be inserted in the ear canal, an output transducer, and an ITE microphone housing accommodating at least one ITE microphone, wherein the ITE microphone housing is configured to be positioned in an outer ear, wherein the processor is further configured for processing an audio signal from the at least one ITE microphone and an audio signal from the at least one BTE sound input transducer in such a way that the hearing loss compensated output signal substantially preserves spatial cues.

Abstract: A method and auditory device for automatic evaluation of an input signal for use in an auditory device, the method including the steps of: detecting a signal; processing the signal to determine one or more shape parameters relevant to the change of spectral shape over time of said signal, and the signal level; and on the basis of the shape parameter and the signal level, and a predetermined set of rules, evaluating whether said signal is a useful input signal for said device.

Abstract: There are provided a human body sound transmission system and a method. The human body sound transmission system according to an aspect of the present invention directly or indirectly contacts a human body to apply coupled signals of high frequency sound signals with high frequency signals, which recover sound signals from the high frequency sound signals, to the human body and transmit the sound signals using the human body as a transmission medium, wherein a magnitude of the sound signals recovered around a human's ear is increased in proportion to the number of signals transmitted through the human body by transmitting the plurality of signals including the same sound signals through the human body.

Abstract: A system of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using a transform domain adaptive filter including two or more eigenvalues to measure an acoustic feedback path from the receiver to the microphone, analyzing a measure of eigenvalue spread against a predetermined threshold for indication of entrainment of the transform domain adaptive feedback cancellation filter, and upon indication of entrainment of the transform domain adaptive feedback cancellation filter, modulating the adaptation of the transform domain adaptive feedback cancellation filter.

Abstract: A digital, three-level output driver (7) of the H-bridge variety for a hearing aid (20) obtains a reduced capacitive interference by providing a primary voltage (3) and a secondary voltage (8) for the output driver (7) and applying the secondary voltage (8) to both sides of the output driver (7) whenever the middle level of the three-level output driver (7) is present in the input signal for the output driver (7). The output driver (7) may be controlled from a pulse-width modulated signal, a sigma-delta pulse-density modulated signal, or a combination of those signals. The output driver (7) produces a clocked output signal consisting of a positive level, a negative level, and a zero level for driving an acoustic output transducer of the hearing aid (20). The invention provides a hearing aid (20) and a method of driving an output stage (7) of a hearing aid (20).

Abstract: Artifacts are reduced during a frequency compression of an audio signal in a hearing device, in particular in a hearing aid. An amplitude information of a source channel is obtained, for example a spectral envelope, from several frequency channels of the audio signal. An amplitude corresponding to the amplitude information is then applied on a signal in a target channel of the several frequency channels, on which the source channel is represented during frequency compression.

Abstract: A hearing aid tailored to a hard-of-hearing person is designed using sampled-data control theory. The method is for designing an audio signal processing system for a hearing aid, wherein the system comprises an AD converter for converting an analog audio input signal (yc) inputted to the hearing aid into a digital audio input signal, a hearing aid digital filter (K(z)) for performing a signal processing on the digital audio input signal outputted from the AD converter, and a DA converter for converting a digital signal outputted from the hearing aid digital filter into an analog audio output signal to be outputted to the hard-of-hearing person.