Abstract: A method (300) of operating a hearing aid with a low delay beam former and a hearing aid (200, 500).

Abstract: An electronic device includes a sensor; a memory storing instructions; and a processor configured to execute the instructions to: estimate a field of view (FOV) of a user by using another sensor included in a wireless earphone; estimate a FOV of the electronic device by using the sensor; compare the estimated FOV of the user with the estimated FOV of the electronic device; determine whether the user gazes at a screen of the electronic device based on the comparison result; recognize a gaze of the user based on determining that the user gazes at the screen of the electronic device; and perform a specified function based on the gaze of the user.

Abstract: A system may include a wearable camera configured to capture images and a microphone configured to capture sounds. The system may also include a processor programmed to receive audio signals from the microphone and detect, based on analysis of the audio signals, a first audio signal associated with a first time period. The first audio signal may be representative of a voice of a single individual. The processor may also be programmed to detect, based on analysis of the audio signals, a second audio signal associated with a second time period. The second time period may be different from the first time period, and the second audio signal may be representative of overlapping voices of two or more individuals. The processor may further be programmed to selectively condition the first and second audio signals, and cause transmission of the conditioned first audio signal to a hearing interface device.

Abstract: Hearing instruments, such as hearing aids, may improve a quality of presented audio through the use of a binaural application, such as beamforming. The binaural application may require communication between the hearing instruments so that audio from a left hearing instrument may be combined with audio from a right hearing instrument. The combining at a hearing instrument can require synchronizing audio sampled locally with sampled audio received from wireless communication. This synchronization may cause a noticeable delay of an output of the binaural application if the latency of the wireless communication is not low (e.g., a few samples of delay). Presented herein is a low-latency communication protocol that communicates packets on a sample-by-sample basis and that compensates for delays caused by overhead protocol data transmitted with the audio data.

Abstract: Disclosed is a digital hearing device with a microphone in an ear band, the digital hearing device including a body; an ear band configured to extend from one side of the body and to surround at least a portion of auricle of a user; a plurality of microphones provided to the ear band at predetermined intervals; an ear head configured to protrude inward from the body and to fit into an ear of the user; and a speaker provided to the ear head and configured to amplify sound received from the plurality of microphones and to deliver the amplified sound to the user.

Abstract: A method (600) of operating a binaural ear level audio system in order to improve speech intelligibility and a binaural ear level audio system adapted to carry out said method.

Abstract: Embodiments of this disclosure provide a mobile terminal, and relate to the field of communications technologies. The mobile terminal includes a screen, an earpiece, and a housing, where the screen is mounted on the housing, the earpiece is disposed in the housing, and the screen is provided with a sound output hole. The sound output hole corresponds to a first front cavity sound outlet and a second front cavity sound outlet that are independently arranged in the housing, the first front cavity sound outlet is connected to the earpiece through a first sound guide channel, and the second front cavity sound outlet is connected to the earpiece through a second sound guide channel A difference between a length of the first sound guide channel and that of the second sound guide channel is within a preset range.

Abstract: Aspects describe a dual-planar retaining piece for stabilizing and securing earpiece in a wearer's ear. The retaining piece is either fixed or removable from the earpiece. The retaining piece includes a first cantilevered portion shaped to flexibly fit under the antitragus of a wearer's ear when the earpiece is worn, a second cantilevered portion shaped to flexibly fit under the antihelix of the wearer's ear when the earpiece is worn, and at least one attachment feature that couples the retaining piece to a body of the earpiece, wherein the body is shaped to fit in the lower concha of the wearer's ear when the earpiece is worn. In aspects, the first and second cantilevered portions are integrally formed.

Abstract: A system for selectively amplifying audio signals may include a microphone configured to capture sounds from an environment of a user. The system may also include a processor programmed to: receive audio signals representative of the sounds captured by the microphone; cause selective conditioning of at least one audio signal received by the microphone from a region associated with the recognized individual; and cause transmission of the at least one conditioned audio signal to a hearing interface device configured to provide sound to an ear of the user.

Abstract: An acoustic impedance measuring system configured to measure an acoustic impedance of an acoustic component includes a first chamber, a second chamber, a first sound pressure sensing device, a second sound pressure sensing device and a sound source. The first sound pressure sensing device is configured to sense a sound pressure in the first chamber. The second sound pressure sensing device is configured to sense a sound pressure in the second chamber. The sound source is connected to the first chamber, wherein the sound source generates a sound propagating towards a first cavity inside the first chamber. The acoustic component is disposed between the first chamber and the second chamber for being measured the acoustic impedance of the acoustic component, and the acoustic impedance of the acoustic component is measured by the first sound pressure sensing device and the second sound pressure sensing device.

Abstract: Provided are an earhole-wearable sound collection device, a signal processing device, and a sound collection method for sound collection at a high S/N ratio, with noise influence being reduced not by a noise reduction process. In the earhole-wearable sound collection device, a microphone that collects emitted speech voice is provided in a space that is substantially sealed off from outside and connects to an ear canal of the wearer (the speaker). With the microphone being located in the space sealed off from outside, emitted speech voice that propagates through the ear canal of the wearer is collected. In a sound collection signal obtained through the ear canal, the emitted speech voice component is dominant over the noise component particularly at low frequencies.

Abstract: A technique for rotational correction of a microphone array includes generating first audio signals representative of sounds emanating from an environment and captured with an array of microphones of an ear-mountable listening device; identifying a characteristic human behavior having at least one of a typical head orientation or a typical head motion associated with the characteristic human behavior by monitoring sensors mounted in fixed relation to the array of microphones; determining a rotational position of the array of microphones relative to the ear based at least in part upon identifying the characteristic human behavior; applying a rotational correction to the first audio signals to generate a second audio signal, wherein the rotational correction is based at least in part upon the rotational position; and driving a speaker of the ear-mountable listening device with the second audio signal to output audio into an ear.

Abstract: Presented herein are techniques for generating a combinatory microphone signal from sounds captured at a microphone array. More specifically, sound signals captured by a microphone array are used to generate first and second directional signals. A cross-power signal is computed from the first and second directional signals. The cross-power signal is converted into an amplitude domain output signal, and a phase of the amplitude domain output signal is reconstructed in order to generate a combinatory microphone signal that is useable for subsequent sound processing operations.

Abstract: A hearing aid and method for use of the same are disclosed. In one embodiment, the hearing includes a body that at least partially conforms to the contours of the external ear and is sized to engage therewith. Various electronic components are contained within the body, including an electronic signal processor that is programmed with a preferred hearing range, which may be an about 10 Hz frequency to an about 30 Hz frequency range of sound corresponding to highest hearing capacity of a patient. Sound received at the hearing aid is converted to the preferred hearing range prior to output.

Abstract: The inventive concept relates to a smart hearing device for providing a control parameter and feedback for a natural language or a non-natural language determined by analyzing sound data, which includes a receiving unit that receives sound data of a voice signal and a noise signal from a first microphone and a second microphone being formed at one side, a determination unit that compares digital flow of the sound data with a previously stored graph pattern to determine a natural language or a non-natural language for the sound data, a processing unit that matches similar data for the determined natural language or non-natural language, based on a database including a natural language area and a non-natural language area, and a providing unit that provides a user with a one-sided sound converted by setting a control parameter in a natural language or a non-natural language specified according to the matched similar data.

Abstract: An over-the-ear hearing assessment device and a method for evaluating the performance of over-the-ear hearing devices are described. An exemplary over-the-ear hearing assessment device includes an ear cup defining an interior volume and positionable at least partially over the ear of a user. The ear cup includes a shell, a cushion, and an acoustic port extending from an exterior to the interior volume of the ear cup. The acoustic port is sealably engagable with a microphone.

Abstract: Various embodiments provide a multi-audio input, stereo audio output, push-to-talk (PTT) switch device. The device may include an audio processing unit configured to perform spatial separation/positioning for one or a plurality of audio sources. The audio processing unit of the device may apply unique head-related transfer functions (HRTFs) to produce left and right audio outputs that correspond to predetermined or dynamically positioned spatial locations for each of the incoming audio streams.

Abstract: 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.

Abstract: Techniques for modifying sound based on a direction of interest include determining a direction of interest associated with a user; receiving a set of audio signals associated with the direction of interest; determining one or more salient frequency bands within the set of audio signals; in response to receiving a command from the user to enhance or suppress the set of audio signals, enhancing or suppressing a portion of the set of audio signals corresponding to the one or more salient frequency bands to create a modified set of audio signals; and outputting the modified set of audio signals.

Abstract: A method and device for automatically increasing the spectral bandwidth of an audio signal including generating a “mapping” (or “prediction”) matrix based on the analysis of a reference wideband signal and a reference narrowband signal, the mapping matrix being a transformation matrix to predict high frequency energy from a low frequency energy envelope, generating an energy envelope analysis of an input narrowband audio signal, generating a resynthesized noise signal by processing a random noise signal with the mapping matrix and the envelope analysis, high-pass filtering the resynthesized noise signal, and summing the high-pass filtered resynthesized noise signal with the input narrowband audio signal. Other embodiments are disclosed.

Abstract: A hearing device includes: a first microphone and a second microphone for provision of a first microphone input signal and a second microphone input signal, respectively; a beamforming module configured to process the first microphone input signal and the second microphone input signal, the beamforming module configured to provide a beamformed input signal; a processor configured to process the beamformed input signal for provision of an electrical output signal based on the beamformed input signal from the beamforming module; a receiver configured to convert the electrical output signal to an audio output signal; and a motion detector; wherein the beamforming module comprises a beamforming controller coupled to the motion detector, and wherein the beamforming controller is configured to control the beamforming module based on motion data from the motion detector.

Abstract: Broadly speaking, the embodiments disclosed herein describe replacing a current hearing aid profile stored in a hearing aid. In one embodiment, the hearing aid profile is updated by sending a hearing aid profile update request to a hearing aid profile service, receiving the updated hearing aid profile from the hearing aid profile service, and replacing the current hearing aid profile in the hearing aid with the updated hearing aid profile.

Abstract: An acoustic device with an open audio device structure that is configured to be carried on the head or upper torso of a user, a housing carried by the open audio device structure, the housing having opposed first and second ends, a flat diaphragm in the housing and comprising a front face and a rear face, the diaphragm configured to radiate front acoustic radiation from its front face and rear acoustic radiation from its rear face, wherein the front and rear acoustic radiations are out of phase, structure that supports the diaphragm such that the diaphragm can move relative to the housing, a primary magnet adjacent to the rear face of the diaphragm, a magnetic circuit that defines a path for magnetic flux of the primary magnet, a voice coil that is exposed to the magnetic flux and is configured to move the diaphragm up and down along a radiation axis that is normal to the front face of the diaphragm, and first and second sound-emitting outlets in the housing, wherein the first sound-emitting outlet is in or p

Abstract: A communication device and method can include one or more processors operatively coupled to memory, a sensor and an output device, where the one or more processors to perform operations of discovering neighboring short range communication enabled devices such as Bluetooth LE devices, creating presence lists from the discovered devices, and transferring biometric and personal data at least to or from the communication device or at least to or from one of the discovered devices. Other embodiments are disclosed.

Abstract: A method for directional signal processing for a hearing aid includes generating first and second input signals from an ambient sound signal using first and second input transducers of the hearing aid and forming first and second directional signals based on the input signals. The directional signals have relative attenuations in directions of first and second useful signal sources. First and second amplification parameters for amplification of first and second useful signals of the signal sources are ascertained. A reference directional characteristic is defined for a reference directional signal.

Abstract: A method of and system for visualizing a sound source is disclosed. The method may include analyzing an audio signal received by a sound transducer to determine a positional direction of the sound source, determining whether the positional direction of the sound source falls outside a field of view of a user, and in response to determining that the positional direction of the sound source falls outside the field of view of the user, rendering on a display unit a visual representation of the sound source. The visual representation of the source is rendered on a virtual surface at a location within the field of view of the user, the location corresponding to at least one of a distance of the source from the user and a positional direction of the source with respect to the user.

Abstract: This disclosure describes an apparatus and method of an embodiment of an invention that is a conferencing device that includes a microphone array that further comprises a plurality of microphones; a processor configured to execute the following steps: performing a beamforming operation to combine the plurality of microphones into a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals, each of the plurality of combined signals corresponding to a different fixed beam; wherein performing the beamforming operation includes applying beamforming weights to the signals from each microphone to achieve a desired pickup pattern that includes a main lobe and sidelobes together with nulls for each fixed beam; performing an acoustic echo cancellation operation generate a plurality of combined echo cancelled signals; and selecting one or more of the combined echo cancelled signals for transmission to the far end.

Abstract: A hearing device comprises an input transducer comprising a microphone for providing an electric input signal representative of sound in the environment of the hearing device, a pre-processor for processing electric input signal and providing a multitude of feature vectors, each being representative of a time segment thereof, a neural network processor adapted to implement a neural network for implementing a detector configured to provide an output indicative of a characteristic property of the at least one electric input signal, the neural network being configured to receive said multitude of feature vectors as input vectors and to provide corresponding output vectors representative of said output of said detector in dependence of said input vectors.

Abstract: A HRTF used for 3D spatialized audio is combined with, e.g., by concatenation, additional settings to provide a more comfortable, accessible, and enjoyable experience for a listener such as a player of a computer game listening to audio through a headset. A single transfer function is thus created that includes the other settings, and once the transfer function is computed the run-time processing can be treated as a single combined transfer function rather than multiple separate stages, resulting in computational savings. The additional settings pertain to hearing aids normally worn by the listener as well as a room-related function specific to a particular listening venue.

Abstract: The present disclosure relates to a cover plate for covering a lateral opening of a shell of an earpiece configured to be worn at least partially within an ear canal, wherein the cover plate comprises a circumferential trench for receiving an antenna on a medial side of the cover plate, wherein the cover plate comprises a circumferential wall radially outward from the trench and adjacent the wall a surface for gluing the shell on.

Abstract: The present application provides a wireless headphone assembly, which includes: a wireless headphone including a left earphone and a right earphone; an adaptor including a plug and a wireless communication module that are electrically connected to each other; the plug is configured for plugging into an interface of the an electronic device, such that the electronic device is in wireless communication with the wireless headphone through the wireless communication module; and a rechargeable earphone box including a box body and a box cover; the box body is provided therein wherein a left earphone receiving groove and a right earphone receiving groove, and the rechargeable earphone box is further provided with an adaptor receiving groove. The wireless headphone assembly provided by the present application can receive the adaptor, such that the adaptor is not easy to lose, and it is convenient for users to use.

Abstract: A hearing device, e.g. a hearing aid, comprises a) a multitude of input units, each providing an electric input signal representing sound in the environment of the user in a time-frequency representation, wherein the sound is a mixture of speech and additive noise or other distortions, e.g. reverberation, b) a multitude of beamformer filtering units, each being configured to receive at least two, e.g. all, of said multitude of electric input signals, each of said multitude of beamformer filtering units being configured to provide a beamformed signal representative of the sound in a different one of a multitude of spatial segments, e.g. spatial cells, around the user, c) a multitude of speech probability estimators each configured to receive the beamformed signal for a particular spatial segment and to estimate a probability that said particular spatial segment contains speech at a given point in time and frequency, wherein at least one, e.g.

Abstract: One or more context aware processing parameters and an ambient audio stream are received. One or more sound characteristics associated with the ambient audio stream are identified using a machine learning model. One or more actions to perform are determined using the machine learning model and based on the one or more context aware processing parameters and the identified one or more sound characteristics. The one or more actions are performed.

Abstract: A system may include a camera configured to capture images from an environment of a user and a microphone configured to capture sounds from an environment of the user. The system may also include a processor programmed to: receive the images; identify a representation of a first individual and a representation of a second individual in the images; receive, from the microphone, a first audio signal associated with a voice of the first individual and a second audio signal associated with a voice of the second individual; detect an amplification criteria indicative of a voice amplification priority between the first individual and the second individual; selectively amplify the first audio signal relative to the second audio signal when the amplification criteria indicates that the first individual has voice amplification priority over the second individual; and cause transmission of the selectively amplified first audio signal to a hearing interface device.

Abstract: The present disclosure relates a directivity hearing-aid device which primarily includes an audio pick-up device, a processor, a selector, and a speaker disposed in a case. The audio pick-up device receives a sound in a range which is pointed by a beam, and the processor generates a detected sound component by amplifying a sound feature point audio in a sound component corresponding to a detecting target which is selected by the selector and generates adjusted sound components by suppressing or shielding the sound feature point audio in the other sound components. Then the processor combines the detected sound component and the adjusted sound components to generate an output sound signal. Accordingly, the directivity hearing-aid device of the present disclosure provides selectively receiving the detected sound component and suppressing or shielding the other impurity sounds to facilitate the user hears audio from a specific object thereby.

Abstract: A system may include a wearable camera configured to capture a plurality of images from an environment of a user and a microphone configured to capture sounds from an environment of the user. The system may also include a processor programmed to receive the images; identify a representation of one individual in one of the images; identify a lip movement associated with a mouth of the individual, based on analysis of the images; receive audio signals representative of the sounds; identify, based on analysis of the sounds, a first audio signal associated with a first voice and a second audio signal associated with a second voice; cause selective conditioning of the first audio signal based on a determination that the first audio signal is associated with the identified lip movement; and cause transmission of the selectively conditioned first audio signal to a hearing interface device.

Abstract: A microphone assembly is provided, comprising a transducer assembly including a first enclosure defining a first acoustic volume and a Micro-Electrical-Mechanical-System (“MEMS”) microphone transducer disposed within the first enclosure. The microphone assembly also includes a second enclosure disposed adjacent to the first enclosure and defining a second acoustic volume in acoustic communication with the first acoustic volume, the second enclosure including an acoustic resistance, wherein the first and second acoustic volumes, in cooperation with the acoustic resistance, create an acoustic delay for producing a directional polar pattern. Circuitry comprising a shelving filter configured to correct a portion of a frequency response of the MEMS microphone transducer is also provided. In some embodiments, the circuitry is embedded within the transducer assembly or at least included within the microphone assembly.

Abstract: Various implementations include control mechanisms for managing hearing aid usage. In some cases, an interface with a representation of the hearing aid in space is used to control audio functions in the device. In other cases, directionality of the device is controlled based upon the user's visual focus direction. In additional cases, the operating mode of the device is adjustable based upon the signature of a nearby acoustic signal.

Abstract: A method of operating a hearing aid system in order to provide improved noise reduction and a hearing aid system (100) adapted to carry out the method.

Abstract: Disclosed are an acoustic module and an electronic product. The acoustic module comprises: a module housing, comprising an upper housing and a lower housing, wherein the upper housing of the module has a side wall and a top wall, the side wall has a through hole thereon, a distance between an upper edge of the through hole and the top wall is greater than or equal to one half of an overall height of the side wall, and the upper housing is capable of being fastened onto the lower housing; a fixing member; and a microphone; wherein the fixing member is fastened and fixed in the upper housing, the fixing member is configured to fix the microphone on an inner surface of the top wall, and the fixing member has a sound channel formed therein through which the microphone and the through hole are connected and are in communication.

Abstract: A method operates an apparatus for tinnitus characterization, in which, in a first step, a broadband test signal having a number of signal frequencies is generated and compared with a tinnitus noise. The respective test signal is stored with an associated comparison result. A probability correlation for determining the tinnitus noise is established based on the stored test signals and comparison results. In a second step, the amplitudes of the individual signal frequencies of the test signal are varied. The first and second steps are performed repeatedly until the probability correlation reaches or exceeds a first threshold value. In a third step, the amplitudes of the individual signal frequencies of the test signal are varied with reference to the probability correlation.

Abstract: A method operates a hearing system which has a hearing instrument worn in an ear. The method includes capturing a sound signal from an environment, processing the captured sound signal in dependence of a set of signal processing parameters, and outputting a processed sound signal to the user. The method further includes analyzing the captured sound signal to recognize own-voice intervals, in which the user speaks, analyzing the captured sound signal for an acoustic feature of the own voice of the user and/or analyzing a signal of a bio sensor. The bio sensor measures at least one non-acoustic vital function of the user and determines, from the acoustic feature and/or the non-acoustic vital function of the user, a measure of an emotional state of the user. A value of a signal processing parameter is adapted, within the own-voice intervals, to change the emotional state if a criterion is fulfilled.

Abstract: An audio enhancement method includes receiving a first plurality of input signals representative of audio captured using an array of two or more sensors, the first plurality of input signals characterized by a first signal-to-noise ratio (SNR), with the audio being the signal-of-interest. The method also includes receiving a second input signal representative of the audio, the second input signal characterized by a second SNR. The second SNR is higher than the first SNR. The method further includes combining the first plurality of input signals and the second input signal to generate one or more driver signals, and driving one or more acoustic transducers using the one or more driver signals to generate an acoustic signal representative of the audio. The driver signals include spatial information derived from the first plurality of input signals, and are characterized by a third SNR that is higher than the first SNR.

Abstract: The present disclosure relates to a method of performing bilateral dynamic range compression of first and second microphone signals generated by first and second hearing aids, respectively, of a binaural hearing aid system. The method comprises to pick-up sound pressure inside an ear canal of the user's left or right ear by a first microphone to generate a first microphone signal in response to incoming sound and pick-up sound pressure inside an ear canal of the user's opposite ear by a second microphone to generate a second microphone signal in response to the incoming sound.

Abstract: A method for a hearing test includes receiving a first testing hearing threshold for a subject at a first frequency under a first background noise level; calculating a first testing signal-to-noise ratio (SNR) between the first testing hearing threshold and the first background noise level; responsive to the first testing SNR, receiving a first adjusted hearing threshold; and responsive to the first adjusted hearing threshold, modifying a first hearing test result of the first frequency.

Abstract: At least one of the first or second hearing prostheses of a binaural hearing prosthesis system includes a dual-mode sound processing unit that is configured to selectively operate in a “sound processing mode” or in a “wireless streaming mode.” When operating in the sound processing mode, the dual-mode sound processing unit is configured to convert received sound signals into output signals for use in stimulating a first ear of a recipient. However, while operating in the wireless streaming mode, the dual-mode sound processing unit is configured to capture input signals (e.g., sound signals, data signals, etc.) and to encode those input signals for direct or indirect wireless transmission to the sound processing unit of the other hearing prosthesis of the binaural hearing system.

Abstract: In a method for the training of a listening situation classifier for a hearing aid, a user is presented with a number of acoustic signals by use of a terminal device, which prompts the user to indicate the signal source of the signal or the particular presented signal. The training data is adapted for the listening situation classifier in dependence on the user's indication of the presented signal or one of the possibly several signals presented and updates the listening situation classifier by use of the training data.

Abstract: Audio enhancement systems, devices, methods, and computer program products are disclosed. In particular embodiments, audio is separated by source at an auxiliary processing device, primary voice presence and/or relevancy per source is determined and used to determine enhancement data that is sent to one or more earpieces and used for enhancing audio at the earpiece. These and other embodiments are disclosed herein.

Abstract: The present disclosure relates to electronic components and glasses comprising an electronic component. The electronic component may include a component body, a first circuit board, a second circuit board, a first microphone element, and a second microphone element. The component body may include a cavity. The first circuit board and the second circuit board may be inclined to each arranged in the cavity. The first microphone element may be arranged on a sidewall, facing the component body, of the first circuit board. The second microphone element may be arranged on a sidewall, facing the component body, of the second circuit board. A first sound conducting hole may be arranged on a sidewall, opposite to the first microphone element, of the component body. The first sound conducting hole may be configured to conduct a sound to the first microphone element. A second sound conducting hole may be arranged on the sidewall, opposite to the first microphone element, of the component body.

Abstract: A method includes receiving multiple frames of time-domain data that includes noise, and computing, for a first frame of the multiple frames, a frequency domain value for each of multiple frequency bins, each frequency bin representing a corresponding range of frequencies. The method also includes determining that a first frequency domain value corresponding to a first frequency bin is less than or equal to a first threshold value, and in response, updating the first frequency domain value based on a function of (i) a smoothing parameter, and (ii) a second frequency domain value corresponding to the first frequency bin. The second frequency domain value is computed using one or more preceding frames of the multiple frames. The method further includes determining a noise floor corresponding to the first frequency bin using the updated first frequency domain value.