Oticon A/s

Sennheiser Communications A/s

Epos Group A/s

Sennheiser Communication A/s

Sennheiser Electronic Gmbh & Co Kg

Svend Feldt

Stig Petri

Michael Syskind Pedersen

Jesper Bruun Jensen

Jan Mark De Haan

Torben Christiansen

Anders Røser Hansen

Poul Peder Hestbek

Casper Fynsk

Mirjana Adnadjevic

Claus Benjaminsen

Esge Brandt Andersen

Nels Hede Rohde

Thomas Bentsen

Anders Brødløs Olsen

Christian Wang

Jakob Sloth Lauridsen

Brian Brorsbøl

Svend Oscar Petersen

Karsten Bo Rasmussen

Troels Holm Pedersen

Thomas Kaulberg

Bjarne Klemmensen

Gary Jones

Tanveer Bhuiyan

Poul Hoang

Anders Vinther Olsen

Nicolai Westergren

Karsten Bonke

Lars Elliot Nyegaard

Andreas Jessen

Sudershan Y S

Jacob Lindvig

Amalie Sofie Ekstrand

Sebastien Curdy-neves

Robert Rehr

Kim Larsen

Mohammad El-sayed

Michael Noes Gätke

Adam Dyrby

Jana Thorin

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

Loudspeaker system provided with dynamic speech equalization

A hearing system comprises a) a multitude of M of microphones providing M corresponding electric input signals x(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 (Δ) of previously determined acoustic transfer function vectors (ATF). The processor is configured A) to determine a constrained estimate of a current acoustic transfer function vector (ATF) in dependence of said M electric input signals and said dictionary (Δ), B) to determine an unconstrained estimate of a current acoustic transfer function vector (ATF) 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.

Hearing system comprising a database of acoustic transfer functions

A hearing device includes a) a multitude of input transducers providing a corresponding multitude of electric input signals; and b) a processor for providing a processed signal in dependence of the electric input signals. The processor includes b1) a beamformer for providing a spatially filtered signal in dependence of electric input signals and beamformer filter coefficients determined in dependence of a fixed steering vector including as elements respective acoustic transfer functions from a target signal source, to each of said multitude of input transducers; and b2) a target adaptation module connected to the input transducers and to at least one beamformer, the target adaptation module being configured to provide compensation signals to compensate the electric input signals so that they match the fixed steering vector.

Hearing device comprising a low complexity beamformer

A hearing device adapted to be worn by a user and for picking up sound containing the user's own voice is provided. The hearing device includes an input unit having first and second input transducers for converting sound to first and second electric input signals, respectively, representing the sound. The hearing device is configured to provide that the first and second input transducers are located on the user so that the user experiences first and second, acoustically different, acoustic environments, respectively, when the user wears the hearing device. The first acoustic environment may be defined as an environment where the own voice signal primarily originates from vibrating parts of the bones and skin or tissue. The second acoustic environment may be defined as an environment where the own voice signal primarily originates from the user's mouth or nose and is transmitted through air from said mouth or nose to the second input transducer(s). A method of operating a hearing device is further disclosed.

Hearing device adapted to provide an estimate of a user's own voice

A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

Differential audio data compensation

A hearing aid system comprises a hearing aid, and a portable auxiliary device' adapted to establish a communication link between them. The hearing aid comprises a microphone providing an electric input signal, a signal processor, and an output unit. The auxiliary device comprises a microphone providing an auxiliary electric input signal, and a user control interface allowing a user to initiate a specific calibration mode of operation of the hearing aid system. The signal processor of the hearing aid is configured to receive corresponding time segments of said electric input signal and said auxiliary electric input signal to provide an estimate of an acoustic transfer function from said microphone of said auxiliary device to said microphone of said hearing aid. A method of operating a hearing aid system is further disclosed. The invention may e.g. be used in various applications related to own voice detection and estimation.

Hearing aid system for estimating acoustic transfer functions

A hearing aid is configured to be located at or in an ear of a user. The hearing aid comprises a heartbeat detector providing a pulse control signal, and a processor. The processor is configured to estimate whether the hearing aid is located at a left or a right ear of the user in dependence of the pulse control signal. A method of operating a hearing aid and a binaural hearing aid system is further disclosed.

Hearing aid comprising a left-right location detector

A hearing device adapted to be worn by a user and for picking up sound containing the user's own voice is provided. The hearing device comprises a) an input unit comprising first and second input transducers for converting sound to first and second electric input signals, respectively, representing said sound; b) a processor configured to receive said first and second electric input signals and to provide a combined signal as a linear combination of the first and second electric input signals, wherein the combined signal comprises an estimate of the user's own voice, and c) wherein said hearing device is configured to provide that said first and second input transducers are located on said user at first and second locations, when worn by said user; and d) wherein said first and second locations are selected to provide that said first and second electric signals exhibit substantially different directional responses for sound from the user's mouth as well as from sound from sound sources located in an environment around the user. A method of operating a hearing device is further disclosed. Thereby an improved quality of an own voice estimate may be provided.

The present invention relates to a communication system for communication of a plurality of stereo audio signals between a plurality of communication devices, wherein the plurality of communication devices comprises a first communication device, a second communication device and at least a third communication device. Each communication device of the plurality of communication devices may comprise a signal processing unit, an audio interface configured to receive a local voice signal of a user of the communication device, a binaural rendering unit configured to render the local voice signal into a stereo local voice signal based on a first spatial information, an input communication interface configured to receive a first stereo audio signal and a second stereo audio signal of the plurality of stereo audio signals transmitted by the second communication device and the third communication device, respectively. The first stereo audio signal may comprise a second voice signal of a second user of the second communication device, and the second voice signal may include a second spatial information, and wherein the second stereo audio signal may comprise a third voice signal of a third user of the third communication device, and where the third voice signal may include a third spatial information. Furthermore, the communication device may comprise an output communication interface configured to transmit a third stereo audio signal of the plurality of stereo audio signals comprising the local voice signal provided with the first spatial information to the second communication device and the third communication device. The first stereo audio signal and the second stereo audio signal may be transmitted to the audio interface, and the user of the communication device experiences a virtual sound environment, wherein the second voice signal and the third voice signal is positioned in the virtual sound environment based on the second spatial information and the third spatial information, respectively.

Communication system for communicating audio signals between a plurality of communication devices in a virtual sound environment

The disclosure presents a method and a communication system comprising; an audio gateway including; a first wireless interface configured to transmit and receive an audio signal via a first communication link, a processing unit configured to transfer the audio signal into an audio streaming signal, a second wireless interface configured to transmit the audio streaming signal via a second communication link, a plurality of communication units comprising a master communication unit and at least a first slave communication unit, wherein each of the communication unit comprises: a first wireless interface configured to receive the audio streaming signal via the second communication link and to communicate with a communication unit, a speaker configured to sound the audio streaming signal received from the audio gateway, a memory unit, and wherein the master communication unit is configured to transmit a pairing request via the second communication link to the audio gateway, and the audio gateway transmits then an encryption key to the master communication unit, and the first slave communication unit is configured to request the encryption key by transmitting an encryption request including an identification of the first slave communication unit to the master communication unit via a first internal communication link, and the master communication unit transmits then the encryption key relating to the second communication link to the first slave communication unit based on an access criteria, and the first slave communication unit is then configured to eavesdrop the audio streaming signal being communicated via the second communication link between the audio gateway and the master communication unit.

Communication system including eavesdropping

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f, A(t, . . . , t)) in the frequency domain; to calculate a transformed second audio signal Y(f, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.

Adaptive filter unit for being used as an echo canceller

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Ballerup, Denmark

Svend Feldt