Abstract: The present disclosure regards a hearing device configured to receive acoustical sound signals and to generate output sound signals comprising spatial cues.

Abstract: An audio processing device comprises a) at least one input unit for providing a 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 reduction system configured to: determine a first signal to noise ratio estimate ?(k,n) of said electric input signal, and determine a second signal to noise signal ratio estimate ?(k,n) of said electric input signal from said first signal to noise ratio estimate ?(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein parameters of said smoothing are determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency band indices. The invention may be used in hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: The present invention regards a hearing aid device at least one environment sound input, a wireless sound input, an output transducer, electric circuitry, a transmitter unit, and a dedicated beamformer-noise-reduction-system. The hearing aid device is configured to be worn in or at an ear of a user. The at least one environment sound input is configured to receive sound and to generate electrical sound signals representing sound. The wireless sound input is configured to receive wireless sound signals. The output transducer is configured to stimulate hearing of the hearing aid device user. The transmitter unit is configured to transmit signals representing sound and/or voice. The dedicated beamformer-noise-reduction-system is configured to retrieve a user voice signal representing the voice of a user from the electrical sound signals. The wireless sound input is configured to be wirelessly connected to a communication device and to receive wireless sound signals from the communication device.

Abstract: A system, method, and computer program product for collaborative online gaming, including at least one of providing a central repository master browser system; providing an experience calibrated match-making service; providing a dynamic multiplayer server component auto deployment and aggregation system; providing a lobby centric simultaneous and collaborative client game play launching feature; and providing a video game screen over-layer technology giving users access to a control interface while inside a video game being played.

Abstract: A hearing aid comprises a) first and second microphones b) an adaptive beamformer filtering unit comprising, b1) a memory comprising a first and second sets of complex frequency dependent weighting parameters representing a first and second beam patterns, b3) an adaptive beamformer processing unit providing an adaptation parameter ?opt(k) representing an adaptive beam pattern, b4) a memory comprising a fixed adaptation parameter ?fix(k) representing a third, fixed beam pattern, b5) a mixing unit providing a resulting complex, frequency dependent adaptation parameter ?mix(k) as a combination of said fixed and adaptively determined frequency dependent adaptation parameters ?fix(k) and ?opt(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 ?mix(k).

Abstract: The application relates to a hearing device adapted for being located at or in a first ear of a user, or to be fully or partially implanted in the head at a first ear of a user, the hearing device comprising a first input transducer for converting a first input sound signal from a sound field around the user at a first location, the first location being a location of the first input transducer, to a first electric input signal, the sound field comprising a mixture of a target sound from a target sound source and possible acoustic noise; a transceiver unit configured to receive a first quantized electric input signal via a communication link, the first quantized electric input signal being representative of the sound field around the user at a second location, the first quantized electric input signal comprising quantization noise due to a specific quantization scheme; a beamformer filtering unit adapted to receive said first electric input signal and said quantized electric input signal and to determine be

Abstract: A method of training an algorithm for optimizing intelligibility of speech components of a sound signal in hearing aids, headsets, etc.

Abstract: The present invention regards a hearing aid device at least one environment sound input, a wireless sound input, an output transducer, electric circuitry, a transmitter unit, and a dedicated beamformer-noise-reduction-system. The hearing aid device is configured to be worn in or at an ear of a user. The at least one environment sound input is configured to receive sound and to generate electrical sound signals representing sound. The wireless sound input is configured to receive wireless sound signals. The output transducer is configured to stimulate hearing of the hearing aid device user. The transmitter unit is configured to transmit signals representing sound and/or voice. The dedicated beamformer-noise-reduction-system is configured to retrieve a user voice signal representing the voice of a user from the electrical sound signals. The wireless sound input is configured to be wirelessly connected to a communication device and to receive wireless sound signals from the communication device.

Abstract: An audio processing device comprises a) at least one input unit for providing a 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 reduction system configured to: determine a first signal to noise ratio estimate ?(k,n) of said electric input signal, and determine a second signal to noise signal ratio estimate ?(k,n) of said electric input signal from said first signal to noise ratio estimate ?(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein parameters of said smoothing are determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency band indices. The invention may be used in hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: The application relates to a hearing device, e.g. a hearing aid, adapted for being located at or in an ear of a user, or for being fully or partially implanted in the head of the user. The application further relates to a method of operating a hearing device. The hearing device comprises a) an input unit for providing at least one electric input signal in a frequency sub-band representation comprising a number K of frequency bands, b) a frequency band to channel allocation unit for allocating said K frequency bands to a number N of frequency channels for each of said electric input signals, wherein K>N; c) antenna and transceiver circuitry allowing reception of at least one further electric signal in said N frequency channels from another device, e.g.

Abstract: A hearing aid comprises a resulting beam former (Y) for providing a resulting beamformed signal YBF based on first and second electric input signals IN1 and IN2, first and second sets of complex frequency dependent weighting parameters W11(k), W12(k) and W21(k), W22(k), and a resulting complex, frequency dependent adaptation parameter ?(k). ?(k) may be determined as <C2*·C1>/<(|C2|2>+c), where * denotes the complex conjugation and (·) denotes the statistical expectation operator, and c is a constant, and wherein said adaptive beam former filtering unit (BFU) comprises a smoothing unit for implementing said statistical expectation operator by smoothing the complex expression C2*·C1 and the real expression |C2|2 over time.

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 ?(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ?(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate ?(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.

Abstract: Embodiments relate generally to methods and systems for communicating, storing, and accessing a calibration and test certificate for a gas detector. A calibration and test certificate may verify that the gas detector has been tested and/or calibrated and can be operated in compliance with specific regulations. Embodiments of the disclosure include integrating the calibration certificate into the individual device, where it may be readily viewed by the user, a supervisor, or inspector to verify the compliance of the device. The certificate may be required to be verified when a user takes the gas detector to a worksite before work can begin. Therefore, storing the certificate on the device may simplify the process of accessing the certificate for the user and supervisor.

Abstract: The present disclosure regards a hearing device configured to receive acoustical sound signals and to generate output sound signals comprising spatial cues.

Abstract: The disclosure relates to binaural hearing instruments and more particularly to reduction of processing time required in a binaural hearing aid system. According to the disclosure, there is provided a method comprising mono-directional transmission of data blocks comprising audio and/or information frames from one hearing instrument to the other hearing instrument or vice versa in a binaural hearing aid. According to the disclosure, the direction of transmission is determined by a quantity characterizing the presence of usable information content in the sound signal picked up by the hearing instruments of the binaural hearing aid. It is proposed to use one or more of local SNR, local voice activity detection indication, local level, local speech intelligibility estimate to determine the direction of transmission, although other quantities may be used.

Abstract: A system for enhancement of noisy speech comprises an input unit is configured to subdivide the spectrum of the input signal into a plurality of frequency sub-bands and to provide time-frequency coefficients X(k,m) for a sequence [X(k,m??D+1) . . . X(k,m?)] of observable noisy signal samples for each of said frequency sub-bands, where k and m are frequency and time indices, respectively, and D is larger than 1. The system further comprises enhancement processing unit configured to receive X(k,m) and to provide enhanced time-frequency coefficients ?(k,m), a storage for statistical model(s) of speech and for statistical model(s) of noise, and an optimizing unit configured to provide said enhanced time-frequency coefficients ?(k,m) using said statistical model of speech and said statistical model of noise, while considering said sequence [X(k,m??D+1) . . . X(k,m?)] of observable noisy signal samples.

Abstract: The application relates to a hearing device, e.g.

Abstract: A binaural aid assistance system comprises first and second hearing assistance devices, each being adapted for being located at or in, or fully or partially implanted in the head at one of a left and right ear, respectively, of a user, wherein each of the first and second hearing assistance devices comprises a) a first wireless interface comprising first antenna and transceiver circuitry adapted for establishing a first communication link to the respective other hearing assistance device based on near-field communication; b) a second wireless interface comprising second antenna and transceiver circuitry adapted for establishing a second bi-directional communication link to an auxiliary device based on far-field communication; and wherein the hearing assistance system is configured to transmit an audio signal picked up by a microphone of the hearing assistance device(s) to said auxiliary device.

Abstract: A hearing aid comprises a) first and second microphones b) an adaptive beamformer filtering unit comprising, b1) a memory comprising a first and second sets of complex frequency dependent weighting parameters representing a first and second beam patterns, b3) an adaptive beamformer processing unit providing an adaptation parameter ?opt(k) representing an adaptive beam pattern, b4) a memory comprising a fixed adaptation parameter ?fix(k) representing a third, fixed beam pattern, b5) a mixing unit providing a resulting complex, frequency dependent adaptation parameter ?mix(k) as a combination of said fixed and adaptively determined frequency dependent adaptation parameters ?fix(k) and ?opt(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 ?mix(k).

Abstract: A hearing aid comprises a resulting beam former (Y) for providing a resulting beamformed signal YBF based on first and second electric input signals IN1 and IN2, first and second sets of complex frequency dependent weighting parameters W11(k), W12(k) and W21(k), W22(k), and a resulting complex, frequency dependent adaptation parameter ?(k). ?(k) may be determined as <C2*·C1>/<(|C2|2>+c), where * denotes the complex conjugation and · denotes the statistical expectation operator, and c is a constant, and wherein said adaptive beam former filtering unit (BFU) comprises a smoothing unit for implementing said statistical expectation operator by smoothing the complex expression C2*·C1 and the real expression |C2>2 over time.