A computing system includes a memory and at least one processor. The memory is configured to store motion data indicative of motion of a hearing instrument. The at least one processor is configured to determine, based on the motion data, whether a user of the hearing instrument perceived a sound. The at least one processor is further configured to output data indicating whether the user perceived the sound.

A method for a wireless personal communication using a hearing system with a hearing device comprises: monitoring and analyzing the user's acoustic environment by the hearing device to recognize one or more speaking persons based on content-independent speaker voiceprints saved in the hearing system; and presenting a user interface to the user for notifying the user about a recognized speaking person and for establishing, joining or leaving a wireless personal communication connection between the hearing device and one or more communication devices used by the one or more recognized speaking persons.

An exemplary spatial enhancement system performs frequency- specific localization and speech comprehension enhancement. Specifically, the system receives an audio signal presented to a recipient of a hearing device, and generates, based on the audio signal, a first frequency signal and a second frequency signal. The first frequency signal includes a portion of the audio signal associated with a first frequency range, and the second frequency signal includes a portion of the audio signal associated with a second frequency range. Based on the first and second frequency signals, the system generates an output frequency signal that is associated with the first and second frequency ranges and that is configured for use by the hearing device in stimulating aural perception by the recipient. This generating of the output frequency signal includes processing the first frequency signal to apply a localization enhancement and processing the second frequency signal to apply a speech comprehension enhancement.

A system for communication between one or more remotely controllable devices and a hearing assistance device includes a gateway device. The hearing assistance device detects voice commands issued by its wearer. The gateway device wirelessly communicates with the hearing assistance device, produces one or more control signals based on the voice commands, and routes the one or more control signals to one or more devices selected from the one or more remotely controllable devices according to the voice command.

A hearing device comprises an ITE-part adapted for being located at or in an ear canal of the user comprising a housing comprising a seal towards walls or the ear canal, the ITE part comprising at least two microphones located outside the seal and facing the environment, and at least one microphone located inside the seal and facing the ear drum. The hearing device may comprise a beamformer filter connected to said at least three microphones comprising a first beamformer for spatial filtering said sound in the environment based on input signals from said at least two microphones facing the environment, and a second beamformer for spatial filtering sound reflected from the ear drum based on said at least one electric input signal from said at least one microphone facing the ear drum and at least one of said input signals from said at least two microphones facing the environment.

A method, comprising receiving at least one sound at an electronic device. The at least one sound is enhanced for the at least one user based on a compound metric. The compound metric is calculated using at least two sound metrics selected from an engineering metric, a perceptual metric, and a physiological metric. The engineering metric comprises a difference between an output signal and a desired signal. At least one of the perceptual metric and the physiological metric is based at least in part on input sensed from the at least one user in response to the received at least one sound.

At least one of the first or second hearing prostheses of a binaural hearing prosthesis system includes a dual-mode sound processing unit that is configured to selectively operate in a “sound processing mode” or in a “wireless streaming mode.” When operating in the sound processing mode, the dual-mode sound processing unit is configured to convert received sound signals into output signals for use in stimulating a first ear of a recipient. However, while operating in the wireless streaming mode, the dual-mode sound processing unit is configured to capture input signals (e.g., sound signals, data signals, etc.) and to encode those input signals for direct or indirect wireless transmission to the sound processing unit of the other hearing prosthesis of the binaural hearing system.

A method for acoustic source separation comprises inputting acoustic data from a plurality of acoustic sensors, combined from a plurality of acoustic sources, converting the acoustic data to time-frequency domain data comprising time-frequency data frames, and constructing a multichannel filter for the time-frequency data frames to separate signals from the acoustic sources. The constructing comprises determining a set of de-mixing matrices (W.sub.f) to apply to each time-frequency data frame to determine a vector of separated outputs (y.sub.ft) by modifying each of the de-mixing matrices by a respective gradient value (G;G′) for a frequency dependent upon a gradient of a cost function measuring a separation of the sources by the respective de-mixing matrix. The respective gradient values for each frequency are each calculated from a stochastic selection of the time-frequency data frames.

A hearing aid system comprises a hearing aid configured to be worn on the head at or in an ear of a user. The hearing aid comprises a microphone system comprising a multitude of M of microphones arranged in said hearing aid and adapted to provide M corresponding electric input signals x.sub.m(n), m=1, . . . , M, n representing time. The environment sound at a given microphone comprises a mixture of a) a target sound signal s.sub.m(n) propagated via an acoustic propagation channel from a direction to or a location (θ) of a target sound source to the m.sup.th microphone of the hearing aid when worn by the user, and b) possible additive noise signals v.sub.m(n) as present at the location of the m.sup.th microphone, wherein the acoustic propagation channel is modeled as x.sub.m(n)=s.sub.m(n)h.sub.m(θ)+v.sub.m(n), and wherein h.sub.m(θ) is an acoustic impulse response for sound for that acoustic propagation channel. The hearing aid system comprises A) a processor connected to said number of microphones, and B) a database Θ comprising a multitude of dictionaries Δ.sub.p, p=1, . . . , P, where p is a person index, of vectors, termed ATF-vectors, whose elements ATF.sub.m, m=1, . . . , M, are frequency dependent acoustic transfer functions representing direction- or location-dependent (θ), and frequency dependent (k) propagation of sound from a direction or location (θ) of a target sound source to each of said M microphones, k being a frequency index, k=1, . . . , K, where K is a number of frequency bands, when said microphone system is mounted on a head at or in an ear of a natural or artificial person (p′), and wherein each of said dictionaries Δ.sub.p comprises ATF-vectors for a given person (p) for a multitude of different directions or locations θ.sub.j, j=1, . . . , J, relative to the microphone system. The processor is configured to, at least in a learning mode of operation, determine personalized ATF-vectors (ATF*) for said user based on said multitude of dictionaries Δ.sub.p of said database Θ, said electric input signals x.sub.m(n), m=1, . . . , M, and said model of the acoustic propagation channels. The invention may e.g. be used in beamforming, own voice estimation, own voice detection, keyword detection, etc.

Systems and methods for modifying audio signals based on context may include at least one microphone configured to capture sounds from an environment of a user; and at least one processor. The processor may be programmed to receive an audio signal representative of sounds captured by the at least one microphone; and determine a context associated with the captured sounds based on the audio signal. Subject to the context being included in a set of stored contexts, the processor may be programmed to determine at least one first speaker whose speech is to be amplified; identify at least one first portion of the audio signal associated with the determined at least one first speaker; amplify the at least one first portion of the audio signal; and transmit to a hearing interface device the amplified at least one first portion of the audio signal.