248results about "Hearing aids signal processing" patented technology

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.

A hearing system comprises a hearing device configured to be worn by a wearer, the hearing device comprising one or more microphones, an acoustic transducer, audio circuitry coupled to the one or more microphones and the acoustic transducer, and a controller coupled to the audio circuitry. The controller is configured to detect a wearer interaction with the hearing device, determine a context of the wearer's use of the hearing device, obtain user data for groups of a population of hearing device users based on user interaction with, and context of use of, the users' hearing devices, determine an operational change to the hearing device based on the wearer interaction, the context, and the user data that corresponds to the wearer's interaction and context, and apply the operational change to the hearing device.

A system and associated processes include a left and right eyewear stem, each including a microphone array comprising a plurality of microphones, and a digital hearing aid that receives content from the microphone array and extracts a desired signal from the content, the digital hearing aid applying frequency-dependent gain to the desired signal to compensate for a user's hearing loss profile. The system further includes processes that receive the desired signal modified with the frequency-dependent gains to acoustically render the desired modified signal proximate to the ear of the user without anything physical being placed within an entrance to an ear canal. A rechargeable battery may be included, along with a front face of the eyewear configured to hold a pair of eyewear lenses, where the front face lacks an electrical conductor connecting the right and left eyewear stem.

The disclosure generally relates to a method, system and apparatus to improve a user's understanding of speech in real-time conversations by processing the audio through a neural network contained in a hearing device. The hearing device may be a headphone or hearing aid. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; a neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and a digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

Disclosed are embodiments of a hearing aid. The hearing aid comprises an input unit configured to provide a first digitized signal associated with a first sampling rate and a second digitized signal associated with a second sampling rate. The first digitized signal comprises a plurality of first frames. The second digitized signal comprises a plurality of second frames. The hearing aid determines the first sampling rate as being different from or as being greater than the second sampling rate such that the time duration of each of each first frame is the same as the time duration of each second frame.

The invention relates to the field of chip clamps, in particular to a hearing aid output limit adjusting method, which comprises an adjusting system, and the adjusting system comprises a hardware module and a software module. The hardware module comprises a DSP module, a collection module, a power amplifier module and a power supply module. The software module comprises a signal processing module, a user interface module and a system management module; the DSP module is used for realizing a complex signal processing algorithm and analyzing sound signals in real time through cooperation of an efficient digital signal processor and an artificial intelligence algorithm, and the method is used through modular connection, collects and analyzes user hearing signals in real time, dynamically adjusts output limitation in combination with hearing conditions and environmental factors of a user, and improves the user experience. In this way, comfort of the user in various hearing scenes can be guaranteed, hearing of the user can be effectively protected, perfect balance between the amplification effect and the comfort of the user is achieved, habits of the user are conveniently recorded in cooperation with the DSP module and the system management module, and user experience is improved.

Disclosed herein are embodiments of a method performed by a hearing aid and a hearing aid including one or more processors, a memory, two or more microphones, and an output transducer. The hearing aid is configured to generate a first processed signal (y) based on input signals from the two or more microphones and a steering value, wherein a target direction is associated with the steering value. The steering value can be changed depending on certain criterion.

The application provides a method, device, electronic equipment and readable storage medium for using a hearing aid, the method is applied to the technical field of the hearing aid, and the method comprises the steps of receiving a gesture instruction sent by the hearing aid; determining a target terminal device working mode matched with a target gesture indicated by the gesture instruction according to a preset gesture and a corresponding relationship of the terminal device working mode; and in the case that the target terminal device working mode is a hearing aid volume adjustment mode, adjusting the volume of the hearing aid by performing a posture adjustment operation on the terminal device. The volume of the hearing aid is adjusted through the posture adjustment operation of the terminal device, which not only can reduce the operation of the user and bring a good experience to the user, but also can improve the adjustment efficiency.

A hearing aid comprising an input unit configured to provide first and second digitized signals associated with first and second sampling rates, respectively, the first and second digitized signals comprising a plurality of first frames and a plurality of second frames; a first analysis filter bank configured to generate first frequency domain signals associated with a first frequency range and a second frequency range based on the first digitized signals; a second analysis filter bank configured to generate second frequency domain signals associated with the first frequency range based on the second digitized signals; wherein the first frequency range of the first frequency domain signals and the second frequency domain signals comprise the same frequency band; and a signal processing unit configured to generate a first processed signal based on the first frequency range of the first frequency domain signals and the second frequency domain signals; an output unit configured to output an audible signal based on the first processed signal.

Disclosed herein is a method for determining a characteristic of a hearing instrument, the hearing instrument including at least one input transducer operable to provide an input audio signal responsive to sensing sound in the environment of the hearing instrument, a signal processing unit and at least one output transducer, the method comprising: emitting an acoustic probe signal by the output transducer, receiving an input audio signal from the microphone, analyzing the received input audio signal to determine the characteristic of the hearing instrument from an input transducer response to the emitted acoustic probe signal, wherein the method further comprises filtering the received input audio signal to selectively attenuate one or more signal components corresponding to the acoustic probe signal and wherein emitting the acoustic probe signal comprises emitting a combined acoustic output signal comprising the acoustic probe signal and an acoustic hearing instrument signal obtained from the filtered input audio signal.

The invention discloses a hearing aid speech enhancement method and device based on artificial intelligence, and a medium, and relates to the technical field of speech signal processing, and the method comprises the steps: inputting a standardized speech enhancement input feature into a direction perception high-frequency speech enhancement model, extracting a coding feature through a shared coding part in the direction perception high-frequency speech enhancement model, and carrying out the recognition of the coding feature; determining a target sound source direction through a direction information generation part in a direction perception high-frequency speech enhancement model, converting the target sound source direction into direction-related information, and combining the direction-related information with the encoding features to generate direction enhancement features; and inputting the direction enhancement feature into a high-frequency recovery part of a direction perception high-frequency speech enhancement model, performing enhancement processing on a high-frequency band corresponding to the target sound source to obtain high-frequency enhancement data, and compensating the high-frequency enhancement data according to the audiogram of the wearer to obtain individualized high-frequency enhancement frequency domain amplitude data. According to the invention, the recognition capability of the wearer on the target voice is improved.

The invention relates to a method (24) for operating a hearing aid (2) in which an input signal (16) is generated by means of a microphone (6) based on ambient sound (7). Based on the input signal (16), an output signal (18) is provided, and an output sound (22) is emitted by means of a receiver (10) based on the output signal (18). The unprocessed ambient sound (22) present in the vicinity of the receiver (10) is determined, and the ratio (36) between the unprocessed ambient sound (22) and the output sound (12) is determined for different frequency bands. The amplitude of the output signal (18) for the frequency band is varied, whereby the ratio (36) lies between an upper limit (38) and a lower limit (40). The invention further relates to a hearing aid (2).

A hearing device and related methods are disclosed, the hearing device comprising an interface and one or more processors, the one or more processors comprising a first primary machine learning model and a first secondary machine learning model, the first primary machine learning model for provision of a first primary model output based on a first primary model input, the first secondary machine learning model for provision of a first secondary model output based on a first secondary model input, wherein the first secondary model input is based on the first primary model output.

Disclosed herein are methods for determining a location of a hearing aid, performed by an electronic device having a processor, a memory, and interface circuitry. The method includes obtaining first accelerometer data from a hearing aid. The method includes obtaining second accelerometer data from the electronic device. The method includes determining, based on the first accelerometer data and the second accelerometer data, an acceleration parameter indicative of a difference between the first accelerometer data and the second accelerometer data. In accordance with the acceleration parameter meeting a safety criterion, the method includes generating lost control data and outputting the lost control data to the hearing aid.

The disclosure generally relates to a method, system and apparatus to improve a user's understanding of speech in real-time conversations by processing the audio through a neural network contained in a hearing device. The hearing device may be a headphone or hearing aid. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; a neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and a digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

A hearing aid control method for eliminating an ear blockage effect comprises the following steps: acquiring an audio decibel recorded by a first microphone as a first decibel; the decibel interval where the first decibel falls is judged, the access mode of the microphones is selected based on the decibel interval, and the first access mode is that the first microphone and the second microphone are both electrically connected with the audio access interface, and the valve of the air hole of the hearing aid is in a closed state; the second access mode is that the first microphone is connected with the hearing aid control chip; and when the second access mode is executed, the valve opening degree of the hearing aid air hole is controlled based on the first decibel, and whether the second microphone is electrically connected with the audio access interface is controlled. According to the invention, the automatic opening and closing of the air hole are realized, while the ear plugging effect and the ear canal damp-heat problem are alleviated, the sound quality is ensured and the howling is effectively inhibited through an acoustic management mechanism of software and hardware collaboration, so that the hearing aid can give consideration to both comfort and definition in different use scenes.

The invention provides acoustic equipment and ear-hanging type hearing aid equipment. The acoustic equipment is used for assisting a user in acquiring sound transmitted from a neighborhood along a target direction, and the acoustic sensor module collects external sound from the outside of the acoustic equipment and internal sound emitted from the position of the loudspeaker module to generate a sensor signal; the signal processing circuit performs target operation based on the sensor signal to generate a driving signal; the loudspeaker module receives the driving signal and outputs loudspeaker sound meeting target sensitivity to surrounding sound, the target sensitivity is within a preset auditory frequency range, the acoustic equipment has directivity in distribution of sensitivity to external sound and has the highest value in the target direction, and the loudspeaker sensitivity to internal sound is lower than 3dB. The acoustic equipment not only has a good spatial pointing effect, but also can realize effective suppression of a squeal phenomenon, and provides better acoustic experience for a user.

The application discloses a method for optimizing the working state of a bone conduction earphone, characterized in that the bone conduction earphone comprises an earphone core and at least one vibration sensor, and the method comprises the following steps: acquiring a vibration signal through the at least one vibration sensor, wherein the vibration signal is at least partially derived from the vibration generated by the earphone core in response to an audio signal, and the vibration of the earphone core is configured to be transmitted to a user wearing the bone conduction earphone in a bone conduction manner; determining the vibration response characteristics of the earphone core based on the vibration signal and the audio signal; and feeding back the working state of the bone conduction earphone based on the vibration response characteristics of the earphone core.

A method includes: initializing signal-processing parameters of the hearing aid at different frequencies based on attribute data and hearing characteristic data of a user, wherein the signal-processing parameters comprising a gain; processing a hearing test signal to be played based on initialized values of the gain, and obtaining a processed hearing test signal; playing the processed hearing test signal and determining a hearing test result of the user using the hearing aid; and in a case where the hearing test result indicates that the hearing aid does not meet a preset first hear-aiding requirement, adjusting values of the gain of the hearing aid at different frequencies based on the hearing test result, and continuously iterating a hearing test based on the adjusted values of the gain until the hearing aid meets the preset first hear-aiding requirement. A hearing aid is also provided.

Improved hearing systems and methods are disclosed herein. In one example embodiment, a hearing system includes memory device(s), audio input device(s) configured to receive audio input signals including audio information arising from a plurality of sound sources, audio output device(s), and processing device(s). During an inference mode, the processing device(s) are configured to operate in accordance with the first neural network to generate intermediate output signals that, to a higher degree than in the audio information, reflect or emphasize at least one desired sound source component of the audio information arising from a desired one of the sound sources of the plurality of sound sources determined to be the desired one of the sound sources at least indirectly based upon a first undershot angle evident from the audio information. The audio output device(s) are configured to generate audio output signals based at least indirectly upon the intermediate output signals.

Various implementations include hearing assist devices and systems for processing audio signals. In particular implementations, a process includes receiving an input signal via a microphone; performing a sound pressure level (SPL) shift that decreases a gain of the input signal to generate a gain reduced audio signal; amplifying the gain reduced audio signal using dynamic range compression to generate an amplified audio signal; generating a noise reduced amplified signal using active noise reduction that simultaneously processes the input signal; and outputting the noise reduced amplified signal to an electrodynamic transducer.

The invention relates to an audio processing method, a hearing aid device and a hearing aid system, and the method comprises the steps: obtaining audio frequency domain signals of each microphone in two microphone arrays, receiving pickup direction indication information, and determining a pickup suppression direction according to the pickup direction indication information, performing signal processing on each path of audio frequency domain signal according to each path of audio frequency domain signal and the pickup suppression direction to obtain a first directional enhanced audio signal and a second directional enhanced audio signal; the signal processing comprises performing signal amplitude adjustment according to a pickup suppression direction; the first directional enhanced audio signal is played through the first hearing aid unit, and the second directional enhanced audio signal is played through the second hearing aid unit; the two microphone arrays are respectively located on a first hearing aid unit and a second hearing aid unit of the hearing aid device. According to the method, the pickup direction of the equipment can be adjusted, the directionality and flexibility of audio processing are improved, and the quality of the audio signal obtained through processing is correspondingly improved.

An ear-worn device may include an adaptive feedback filter and control circuitry. The control circuitry may determine a metric value based on coefficients of the adaptive feedback filter at causal delays (longer than minimum electroacoustic propagation time) and acausal delays (shorter than minimum electroacoustic propagation time). The metric may indicate entrainment degree in the adaptive feedback filter. Based on the metric, the control circuitry may adjust filter parameters and / or signal processing operations to mitigate entrainment effects.

The invention relates to a sound signal noise reduction method and a hearing aid. The method comprises the steps that based on a sound signal collected by a microphone, an environment scene corresponding to the sound signal is recognized, a first signal-to-noise ratio and a second signal-to-noise ratio matched with the environment scene are determined, the first signal-to-noise ratio is the signal-to-noise ratio when the maximum speech recognition rate is reached after hearing-aid of a user, and the second signal-to-noise ratio is the signal-to-noise ratio when the maximum speech recognition rate is reached after hearing-aid of the user. The second signal-to-noise ratio is a signal-to-noise ratio when the user reaches a speech recognition threshold after hearing aid, and performing noise reduction processing on the sound signal based on the first signal-to-noise ratio or the second signal-to-noise ratio. By adopting the method, personalized noise reduction can be carried out according to the noise tolerance degree of the user and the environmental condition, and the noise reduction effect is effectively improved.

A method operates a hearing aid, in particular a classic hearing aid. The hearing aid includes an input transducer, a signal processing device, and an output transducer. Wherein an occurrence of a comb filter effect is determined by the signal processing device, and wherein a countermeasure is controlled by the signal processing device as a function of the determined occurrence of the comb filter effect.

A hearing aid configured to be worn by a user is provided. The hearing aid comprises a wireless receiver unit configured to receive, at least when said hearing aid is in a wireless reception mode, a wireless signal from a transmitter of another device or system, to provide an audio input signal based thereon, and to identify said audio input signal as originating from one of a multitude of different types of audio transmitters. The hearing aid comprises a first noise reduction system for reducing noise in at least one electric input signal picked up from an environment of the user and for providing a noise reduced signal. The hearing aid comprises an own voice detector configured to provide an own voice control signal indicative of whether or not or with what probability the user's own voice is present in said sound from the environment of the user. The hearing aid comprises an input gain controller configured to apply an input gain to the noise reduced signal in dependence of said own voice control signal and said type of audio transmitter.

Presented herein are techniques for determining a linguistic context from speech (e.g., a conversation) captured by a system and using the "linguistic context" or "second-order meaning" to initiate / generate an action at / by the system. More specifically, a hearing device (e.g., cochlear implant) or medical device is configured to capture / detect speech which has a primary / explicit, or first-order, meaning. An intelligence module that includes an artificial intelligence (AI) model, e.g., a large language model (LLM) or a large action model (LAM), is configured to obtain the speech and determine a linguistic context (second-order meaning) of the speech and determine the relevancy of the second order-meaning to the system and / or to the recipient (recipient-specific second-order meaning of the speech). Using the recipient- specific second-order meaning of the speech, one or more instructions can be generated and / or produced for implementation at the system.

A hearing device is disclosed. The hearing device comprises an interface and a processor. The hearing device is configured to obtain a microphone input signal from a microphone of the hearing device. The hearing device is configured to obtain a transceiver input signal from a transceiver of the interface. The hearing device is configured to determine a time offset between the microphone input signal and the transceiver input signal. The hearing device is configured to determine a time delay based on the time offset. The hearing device is configured to process the transceiver input signal for providing an output signal, wherein processing the transceiver input signal comprises applying the time delay to the transceiver input signal for providing a delayed transceiver input signal. The hearing device is configured to provide the output signal to a receiver of the hearing device for providing an audio output signal.

The invention relates to a method (24) for operating a hearing device (2), in which an input signal (16) is generated by means of a microphone (6) as a function of ambient sounds (7). An output signal (18) is provided as a function of the input signal (16), and an output sound (22) is output as a function of the output signal (18) by means of the earpiece (10). An unprocessed ambient sound (22) present within the range of the earpiece (10) is determined, and a ratio (36) between the unprocessed ambient sound (22) and the output sound (12) is determined for different frequency bands. The amplitude of the output signal (18) of the associated frequency band is varied, wherein the ratio (36) is set between an upper limit value (38) and a lower limit value (40). The invention also relates to a hearing device (2).