Abstract:
The application relates to a binaural hearing system comprising left and right hearing devices, e.g. hearing aids, each comprising a) a multitude of input units, each providing a time-variant electric input signal xi(t) representing sound received at an ith input unit, t representing time, the electric input signal xi(t) comprising a target signal component si(t) and a noise signal component vi(t), the target signal component originating from a target signal source; b) a configurable signal processing unit for processing the electric input signals and providing a processed signal y(t); c) an output unit for creating output stimuli to the user, d) transceiver circuitry allowing information to be exchanged between the hearing devices, and e) a binaural speech intelligibility (SI) prediction unit for providing a binaural SI-measure of the predicted speech intelligibility of the user when exposed to said output stimuli, based on processed signals yl(t), yr(t) from the signal processing units of the respective left and right hearing devices. This allows the hearing devices to control the processing of the respective electric input signals based on said binaural SI-measure.
Abstract:
The application relates to a hearing device comprising a beamformer of the generalized sidelobe canceler (GSC) type. The application further relates to a method of operating a hearing device. The disclosure addresses a problem which occurs when using a GSC structure in a hearing device application. The problem arises due to a non-ideal target-cancelling beamformer. As a consequence, a target signal impinging from the look direction can—unintentionally—be attenuated by as much as 30 dB. To resolve this problem, it is proposed to monitor the difference between the output signals from the all-pass beamformer and the target-cancelling beamformer to control a time-varying regularization parameter in the GSC update. This has the advantage of providing a computationally simple solution to the non-ideality of the GSC beamformer. The invention may e.g. be used in hearing aids, headsets, ear phones, active ear protection systems, or combinations thereof.
Abstract:
A hearing assistance system calibrates a noise reduction system of a hearing assistance device. The system comprises a hearing assistance device, and an auxiliary device. The hearing assistance device comprises a multitude of input units, and a multi-channel beamformer filtering unit configured to determine filter weights for a beamformed signal. The system further comprises a user interface for activating a calibration mode. The auxiliary device comprises an output transducer for converting an electric calibration signal to an acoustic calibration sound signal. The system is configured to estimate a look vector for a target signal originating from a target signal source located at a specific location relative to the user based on the acoustic calibration sound signal.
Abstract:
The present invention regards a hearing system configured to be worn by a user comprising an environment sound input unit, an output transducer, and electric circuitry. The environment sound input unit is configured to receive sound from the environment of the environment sound input unit and to generate sound signals representing sound of the environment. The output transducer is configured to stimulate hearing of a user. The electric circuitry comprises a spatial filterbank. The spatial filterbank is configured to use the sound signals to generate spatial sound signals dividing a total space of the environment sound in subspaces. Each spatial sound signal represents sound coming from a subspace. The subspaces may (in particular modes of operation) be either fixed, or dynamically determined, or a mixture thereof.
Abstract:
An improved scheme for identifying and removing musical noise in an audio processing device can be achieved in an audio processing device that includes an analysis path with a perceptive model of the human auditory system and provides an audibility measure. The device may also identify an artifact introduced into the processed signal by the processing algorithm and provide an artifact identification measure, and control a gain applied to a signal of the forward path by the processing algorithm based on the perceptive model. An advantage is to dynamically optimize noise reduction with a view to audibility of artifacts and is applicable to hearing aids, headsets, ear phones, active ear protection systems, hands-free telephone systems, mobile telephones, teleconferencing systems, public address systems, karaoke systems, and classroom amplification systems.
Abstract:
A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters hm(n), whose impulse responses are modulated from a first prototype filter h(n), where m=0, 1, . . . , M−1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.
Abstract:
A hearing device adapted for being located at or in an ear of a user, or for being fully or partially implanted in the head of a user comprises a) an input unit for providing at least one electric input signal representing sound in an environment of the user, said electric input signal comprising a target speech signal from a target sound source and additional signal components, termed noise signal components, from one or more other sound sources, b) a noise reduction system for providing an estimate of said target speech signal, wherein said noise signal components are at least partially attenuated, and c) an own voice detector for repeatedly estimating whether or not, or with what probability, said at least one electric input signal, or a signal derived therefrom, comprises speech originating from the voice of the user. The noise signal components are identified during time segments wherein the own voice detector indicates that the at least one electric input signal, or a signal derived therefrom, originates from the voice of the user, or originates from the voice of the user with a probability above an own voice presence probability (OVPP) threshold value. A method of operating a hearing device is further disclosed.
Abstract:
A hearing aid adapted for being worn by a user comprises at least two microphones, providing respective at least two electric input signals representing sound; a filter bank converting the at least two electric input signals into signals as a function of time and frequency; a directional system connected to said at least two microphones and being configured to provide a filtered signal in dependence of said at least two electric input signals and fixed or adaptively updated beamformer weights. At least one direction to a target sound source is defined as a target direction. For each frequency band, one of said at least two microphones is selected as a reference microphone, thereby providing a reference input signal for each frequency band. The reference microphone for a given frequency band may be selected in dependence of directional data related to directional characteristics of the at least two microphones.
Abstract:
A hearing system comprises a) a multitude of M of microphones providing M corresponding electric input signals xm(n), m=1, . . . , M, and n representing time, b) a processor connected to said multitude of microphones and providing a processed signal in dependence thereof, c) an output unit for providing an output signal in dependence of said processed signal, and d) a database (Θ) comprising a dictionary (Δpd) of previously determined acoustic transfer function vectors (ATFpd). The processor is configured A) to determine a constrained estimate of a current acoustic transfer function vector (ATFpd,cur) in dependence of said M electric input signals and said dictionary (Δpd), B) to determine an unconstrained estimate of a current acoustic transfer function vector (ATFuc,cur) in dependence of said M electric input signals, and C) to determine a resulting acoustic transfer function vector (ATF*) for a user of the hearing system in dependence thereof and of a confidence measure related to said electric input signals. A method of operating a hearing device is also disclosed. Thereby an improved noise reduction system for a hearing aid or headset may be provided.
Abstract:
A hearing aid comprises at least one input unit for providing at least one stream of samples of an electric input signal in a first domain; at least one encoder configured to convert said at least one stream of samples of the electric input signal in the first domain to at least one stream of samples of the electric input signal in a second domain; a processing unit configured to process said at least one electric input signal in the second domain, to provide a compensation for the user's hearing impairment, and to provide a processed signal as a stream of samples in the second domain; a decoder configured to convert said stream of samples of the processed signal in the second domain to a stream of samples of the processed signal in the first domain. The at least one encoder is configured to convert a first number of samples from said at least one stream of samples of the electric input signal in the first domain to a second number of samples in said at least one stream of samples of the electric input signal in the second domain. The decoder is configured to convert said second number of samples from said stream of samples of the processed signal in the second domain to said first number of samples in said stream of samples of the electric input signal in the first domain. The second number of samples is larger than the first number of samples. The at least one encoder is trained, and at least a part of said processing unit providing said compensation for the user's hearing impairment is implemented as a trained neural network. A method of operating a hearing aid is further disclosed. Thereby an improved hearing aid may be provided.