118results about "Sets with desired directivity" patented technology

A system for separation of a user's voice from ambient sound, comprising a device to be worn at the user's ear or at least partly in the user's ear canal comprising a first microphone oriented acoustically outwardly towards the environment and a second microphone oriented acoustically inwardly towards the user's ear canal, and an audio signal processing unit for processing audio signals from the first and second microphone by a blind source separation algorithm adapted to separate the user's voice from ambient sound.

A hearing aid system includes a first microphone and a second microphone for provision of electrical input signals, a beamformer for provision of a first audio signal based at least in part on the electrical input signals, the first audio signal having a directional spatial characteristic, wherein the beamformer is configured to provide a second audio signal based at least in part on the electrical input signals, the second audio signal having a spatial characteristic that is different from the directional spatial characteristic of the first audio signal, and a mixer configured for mixing the first audio signal and the second audio signal in order to provide an output signal to be heard by a user.

The present invention is adapted to assist in optimizing the performance of a hearing aid by detecting conditions which result in sub-optimal performance and assisting a user or health care professional to correct those conditions. The present invention is adapted to optimize the performance of a hearing aid by generating a reference signal which, when broadcast, creates a feedback signal. The feedback signal is then measured, or its characteristics identified, to determine whether the hearing aid system is optimized and, if not, what components or characteristics are not optimized and how to optimize the hearing aid system performance or that of its components.

A hearing aid that analyzes a surrounding acoustic environment and automatically switches between a plurality of hearing aid processing reduces noise by limiting directionality, when the user is in a noisy outdoor location. However, in the case where directionality is limited to the front when the user is walking or the like, the user is put in extreme danger because he/she cannot notice sound of danger approaching from behind. Behavior analysis of identifying a walking state of the user is necessary in addition to environmental analysis, but typical walking detection using a sensor as in the case of a pedometer and the like is not applicable to a device worn at an ear such as a hearing aid. On the basis of an occurrence pattern of wind noise when walking, the walking state of the user is identified in the case where pulse-like wind noise occurs repeatedly. This enables walking detection to be performed using an existing structure, with there being no need to provide a sensor or the like. Hence, it is possible to provide a hearing aid that can be safely used even outdoors.

The invention discloses a deep learning-based binaural sound source positioning method in a digital hearing aid. The method comprises the following steps: firstly, decomposing a binaural sound sourcesignal into a plurality of channels through a gammatone filter, extracting a high-energy channel through a weighting coefficient, then extracting a first type of features by using a head-related-transform function (HRTF), namely an Interaural Time Difference (ITD) and an Interaural Intensity Difference (IID) which are used as inputs of deep learning, and dividing a horizontal plane into four quadrants to narrow a positioning range; secondly, extracting a second type of features of head-related transform, namely an Interaural Level Difference (ILD) and an Interaural Phase Difference (IPD); andfinally, in order to realize more precise positioning, taking the first and second types of four features as inputs of next deep learning, thereby obtaining an azimuth angle of sound source positioning. Precise positioning of 72 azimuth angles is realized on the horizontal plane from 0 degree to 360 degrees at a step length of 5 degrees.

The hearing aid apparatus includes a first hearing aid member and a second hearing aid member. The first hearing aid member is placeable on a patient's body, and includes a first transducer for receiving sounds that would be received by the patient's first ear and converting these received sounds into first transmittable signals. The second hearing aid is placeable on a patient's body adjacent to the patient's second ear, and includes a receiver for receiving the first transmittable electrical signals and a second transducer for converting the first transmittable electrical signals into sound signals configured for delivery to the patient's second ear.

Systems and methods for performing source separation are provided. Source separation is performed using a composite signal and a signal dictionary. The composite signal is a mixture of sources received by a sensor. The signal dictionary is a database of filtered basis functions that are formed by the application of directional filters. The directional filters approximate how a particular source will be received by the sensor when the source originates from a particular location. Each source can be characterized by a coefficient and a filtered basis function. The coefficients are unknown when the sources are received by the sensor, but can be estimated using the composite signal and the signal dictionary. Various ones of the sources may be selectively reconstructed or separated using the estimated value of the coefficients.

A speech intelligibility enhancing system for difficult acoustical conditions is disclosed, the speech intelligibility enhancing system comprising at least one ear plug (201) for insertion in an ear canal (218) of a person, the at least one ear plug being arranged with an ear canal facing portion (401) and an environment facing portion (402), and the at least one ear plug comprising an acoustically attenuating path (214; 214, 213) comprising a vent (214) coupling said environment facing portion (402) with said ear canal facing portion (401); and an electroacoustic path (202, 204, 209; 202, 203, 204, 208, 209, 210, 211, 212) comprising a microphone (202) at said environment facing portion (402), a variable gain (204) and a loudspeaker (209) at said ear canal facing portion (401); wherein said acoustically attenuating path (214; 214, 213) is arranged with a transfer function from said environment facing portion (402) to said ear canal facing portion (401) having a low pass characteristic having a low pass cut¬off frequency and said low pass characteristic attenuating sound by a nominal attenuation (Go) for frequencies below said cut-off frequency.

The application discloses direction of arrival estimation in a miniature device using a sound sensor array. The miniature device is a hearing device which includes a sound system for estimating the direction of arrival of acoustical signals emitted by one or more sound sources, wherein the system includes a sound sensor unit including an array of M sound receiving transducers, each providing an electric input signal; a sampling unit used for providing at least one sample of a surrounding sound field; and a processing unit comprising a model unit comprising a parametric model configured to be able to describe the sound field at the array as a function of the direction of arrival in a region surrounding and adjacent to the array, a model optimizing unit configured to optimize said model based on said sound samples, a cost optimizing unit configured to minimize a cost function of the model with respect to said direction of arrivals, and an estimating unit configured to estimate the direction of arrival based on said parametric model with the optimized parameters and the optimized cost function.

An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.