Abstract:
A hearing aid system comprising a pair of hearing devices, e.g. hearing aids, worn at the ears of a user receives a target signal generated by a target signal source and transmitted through an acoustic channel to microphones of the hearing aid system. Due to (potential) additive environmental noise, a noisy acoustic signal is received at the microphones of the hearing system. An essentially noise-free version of the target signal is simultaneously transmitted to the hearing devices of the hearing system via a wireless connection. Based on a sound propagation model of the acoustic propagation channel from the target sound source to the microphones of the hearing aid system, and on relative transfer functions representing direction-dependent filtering effects of the head and torso of the user in the form of direction-dependent acoustic transfer functions from a microphone on one side of the head, to a microphone on the other side of the head, a direction-of-arrival (DoA) of the target sound signal relative to the user is determined using a maximum likelihood approach.
Abstract:
An audio processing device comprises a) at least one input unit for providing time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise detection and/or reduction system configured to b1) determine an a posteriori signal to noise ratio estimate γ(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ζ(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate γ(k,n) based on a recursive decision directed algorithm. The application further relates to a method of of estimating an a priori signal to noise ratio. The invention may e.g. be used for the hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.(Fig. 1A should be published)
Abstract:
A hearing aid comprises a) first and second microphones b) an adaptive beamformer filtering unit comprising, b1) a first and second memories comprising a first and second sets of complex frequency dependent weighting parameters representing a first and second beam patterns, where said first and second sets of weighting parameters are predetermined initial values or values updated during operation of the hearing aid, b3) an adaptive beamformer processing unit providing an adaptation parameter βopt(k) representing an adaptive beam pattern configured to attenuate unwanted noise under the constraint that sound from a target direction is essentially unaltered, b4) a third memory comprising a fixed adaptation parameter βfix(k) representing a third, fixed beam pattern, b5) a mixing unit providing a resulting complex, frequency dependent adaptation parameter βmix(k) as a combination of said fixed and adaptively determined frequency dependent adaptation parameters βfix(k) and βopt(k), respectively, and b6) a resulting beamformer (Y) for providing a resulting beamformed signal YBF based on first and second microphone signals, said first and second sets of complex frequency dependent weighting parameters, and said resulting complex, frequency dependent adaptation parameter βmix(k).
Abstract:
An intrusive binaural speech intelligibility predictor system receives a target signal comprising speech in left and right essentially noise-free and noisy and/or processed versions at left and right ears of a listener. The system comprises a) first, second, third and fourth input units for providing time-frequency representations of said left and right noise-free and noisy/processed versions of the target signal, respectively; b) first and second Equalization-Cancellation stages adapted to receive and relatively time shift and amplitude adjust the left and right noise-free and noisy/processed versions, respectively, and to provide resulting noise-free and noisy/processed signals, respectively; and c) a monaural speech intelligibility predictor unit for providing final binaural speech intelligibility predictor value SI-Measure based on said resulting noise-free and noisy/processed signals. The Equalization-Cancellation stages are adapted to optimize the SI-Measure to indicate a maximum intelligibility of said noisy/processed versions of the target signal by said listener. The invention may e.g. be used in development systems for hearing aids.
Abstract:
The present disclosure regards a hearing device comprising a power source, electric circuitry, a loudspeaker, at least one microphone for sound from an acoustic environment, and at least one wireless receiver for wirelessly received sound signals. The microphone is configured to generate an environment sound signal. The wireless receiver is configured generate a source sound signal. The electric circuitry is configured to estimate at least one parameter of an impulse response from the location of the origin of the wirelessly received signal to the location of a user of the hearing device in dependence on the source sound signal and the environment sound signal. The electric circuitry is further configured to process the environment sound signal in dependence on the estimated at least one impulse-response parameter, thereby generating an output sound signal. The output sound signal is processed into sound by the loudspeaker.
Abstract:
An audio processing device comprises a forward path comprising an input unit for delivering a time varying electric input signal representing an audio signal, the electric input signal comprising a target signal part and a noise signal part, a signal processing unit for processing said electric input signal and providing a processed signal, and an output unit for delivering an output signal based on said processed signal. An audio processing device comprises an analysis path comprising a model unit comprising a perceptive model of the human auditory system and providing an audibility measure, an artifact identification unit for identifying an artifact introduced into the processed signal by the processing algorithm and providing an artifact identification measure, and a gain control unit for controlling a gain applied to a signal of the forward path.