47results about "Occlusion effect electronic compensation" patented technology

A wearable device may be worn in an ear of a user. The wearable device may include a biometric sensor that collects biometric information from the ear, and an environmental sensor that collects environmental information from the ear. The wearable device may receive the biometric information from the biometric sensor, and the environmental information from the environmental sensor, and send biofeedback determined from the biometric information, and an environmental condition determined from the environmental information, wirelessly to a communication device. Other embodiments are disclosed.

Embodiments herein relate to assistive listening devices and systems for providing audio streams to device wearers within sound fields. In an embodiment an assistive listening device is included having a control circuit, an electroacoustic transducer for generating sound in electrical communication with the control circuit, a power supply circuit in electrical communication with the control circuit, and a communications circuit in electrical communication with the control circuit. The control circuit can be configured to issue a communication to an audio communication device or audio provisioning device including at least one of a language preference, a set of hearing requirements, data regarding a presentation delay, and an authorization status identifier, digital code, digital token, or digital key specific to a wearer of the assistive listening device. Other embodiments are also included herein.

An earbud assembly presents audio content to an ear canal of a user. The audio content may be based in part on sounds in a local area surrounding the user. The earbud assembly detects sounds in the area surrounding the user through one or more acoustic sensors. Sound waves pass through holes in a body of the earbud assembly and propagate to a waveguide leading to the one or more acoustic sensors. The earbud assembly processes the detected sound data in a controller, which instructs a speaker assembly to present the audio content based in part on the detected sound data. When presented by the speaker assembly, the detected sound may be amplified, attenuated, filtered, and / or enhanced.

A noise reduction system including a feedforward sensor, an audio controller, and an acoustic driver is provided. The feedforward sensor is arranged to detect body conducted vibrations. The feedforward sensor is configured to generate a feedforward signal based on the detected vibrations. The audio controller is communicatively coupled to the feedforward sensor. The audio controller is configured to generate an audio output signal based on the feedforward signal and a command signal. The acoustic driver is configured to render audio based on the audio output signal. In some examples, the noise reduction system further includes a feedback sensor arranged to capture sound within an ear canal of a user. The feedback sensor is configured to generate a feedback signal based on the captured sound. The audio output signal is generated further based on the feedback signal.

Methods, systems, computer-readable media, and apparatuses for audio signal processing are presented. An apparatus for audio signal processing includes a memory configured to store instructions and a processor configured to execute the instructions. The instructions, when executed, cause the processor to receive an external microphone signal from a first microphone and generate a transmissive component based on the external microphone signal and hearing compensation data. The hearing compensation data is based on a hearing map of a particular user. The instructions, when executed, also cause the processor to cause a speaker to generate an audio output signal based on the transmissive component.

In an active noise reducing headphone, a signal processor applies filters and control gains of both the feed-forward and feedback active noise cancellation signal paths. The signal processor is configured to apply first feed-forward filters to the feed-forward signal path and apply first feedback filters to the feedback signal path during a first operating mode providing effective cancellation of ambient sound, to apply second feed-forward filters to the feed-forward signal path during a second operating mode providing active hear-through of ambient sounds with ambient naturalness, and to apply third feed-forward filters to the feed-forward signal path during a third operating mode providing active hear-through of ambient sounds with a different total response than is provided in the second operating mode.

An earpiece includes: a first end; a second end opposite from the first end; a first channel extending from a first location that is closer to the first end than to the second end, to a second location that is closer to the second end than to the first end; and a first diaphragm, wherein the first diaphragm has a first surface and a second surface opposite the first surface, the first surface of the diaphragm configured to be in fluid communication with a lumen in the first channel, wherein the first diaphragm extends in a direction that is parallel to, or that forms an acute angle with, a longitudinal axis of the first channel.

A manufacturing method for a device includes: providing a wafer including a first layer and a second layer; forming and patterning an actuating material formed on the wafer; patterning the first layer of the wafer to form a trench line; and removing a first part of the second layer. The first layer forms a film structure including a membrane. A slit is formed within and penetrates through the membrane because of the trench line. The film structure is actuated to form a vent temporarily because of the slit. An ear canal and an ambient of a wearable sound device are to be connected via the vent temporarily opened. The slit divides the membrane into a first membrane portion and a second membrane portion. A difference between the displacements of these two membrane portions is larger than a thickness of the membrane when the vent is formed.

In the inventive method for processing ambient sound, which is detected by an audio device (10) worn on or in the ear, a reference signal (S1) is generated from the detected ambient sound. A measurement signal (S1) is generated from the sound detected in the ear canal of a user of the audio device worn on or in the ear. The reference signal is processed into a first signal and the measurement signal into a second signal (S2). A factor is determined based on the first and second signals (S3) so that a measure of the signal strength of the second signal is adjusted to a measure of the signal strength of the first signal. A third signal based on the reference signal is multiplied by the determined factor (S4). The third signal multiplied by the determined factor is output (S5).

A noise reduction system including a feedforward sensor, an audio controller, and an acoustic driver is provided. The feedforward sensor is arranged to detect body conducted vibrations. The feedforward sensor is configured to generate a feedforward signal based on the detected vibrations. The audio controller is communicatively coupled to the feedforward sensor. The audio controller is configured to generate an audio output signal based on the feedforward signal and a command signal. The acoustic driver is configured to render audio based on the audio output signal. In some examples, the noise reduction system further includes a feedback sensor arranged to capture sound within an ear canal of a user. The feedback sensor is configured to generate a feedback signal based on the captured sound. The audio output signal is generated further based on the feedback signal.

Disclosed herein are systems and methods for automatic correction of real ear measurements (REMs). A sound signal is produced through a receiver of a hearing device, and a sound pressure signal is sensed using a microphone placed inside the ear canal. The sound pressure signal is transformed to obtain a frequency response signal, a local minimum of the frequency response signal is detected above a programmable frequency level, and a spectral flatness of the frequency response signal is calculated in a selected frequency band surrounding the local minimum. If the spectral flatness is greater than a selected threshold value, acoustic correction is applied to the frequency response in the selected frequency band using an estimated transfer function to obtain a corrected sound pressure frequency response. The corrected sound pressure frequency response is used to modify, or make a recommendation to modify, a physical or operational characteristic of the hearing device.

A method of generating a suppression signal using an adaptive filter, using an ear canal microphone signal from an earphone and an audio content signal and combining the suppression signal with others signals to generate a mixed signal that is sent to a speaker.

Adaptive occlusion cancellation is performed in an ear-wearable device using an adaptive filter. An adaptive gain of the adaptive filter is used to determine a leakage path estimate between an external source and an eardrum of the user through the ear-wearable device. The leakage path estimate is used to update an adaptive hear-through filter of the ear-wearable device. The updated adaptive hear-through filter is used for hear-through processing in the ear-wearable device.

A method for audio processing in a module to be worn in the ear by a user (202), comprising: Receiving two acoustic signals using two microphones (106, 108), wherein the acoustic signals include a first acoustic signal that is detected outside the ear canal and a second acoustic signal that is detected inside the ear canal; Determine, based at least in part on the acoustic signals, the quality of a seal provided by the module (202) to be worn in the ear when worn by the user; Checking the determined quality of the sealing in relation to a predetermined threshold, and based on the testing: Performing occlusion reduction based on the second acoustic signal to improve sound within the user's ear when the seal quality exceeds the predetermined threshold; and Performing active noise reduction (ANR) based on the first acoustic signal to reduce noise within the user's ear when the sealing quality is below the predetermined threshold.

A method (300) for providing personalized ambient sound playback, ASP, calibration data (303) associated with an audio device and a particular user is presented. The method (300) includes processing (330) a digital representation of a first specific external sound based on an initial set (123) of frequency-dependent processing parameters, and generating (340), by the audio device, a first internal sound (Si) based on the processed digital representation of the first specific external sound (Se) when the audio device is worn by a specific user. The method (300) further comprises adjusting (360) an initial set (123) of frequency-dependent processing parameters based on the acquired first feedback data (301) and providing (370) these parameters as ASP calibration data (303) for the audio device (10) when a particular user (40) wears the audio device (10).

Disclosed herein are systems and methods for automatic correction of real ear measurements (REMs). A sound signal is produced through a receiver of a hearing device, and a sound pressure signal is sensed using a microphone placed inside the ear canal. The sound pressure signal is transformed to obtain a frequency response signal, a local minimum of the frequency response signal is detected above a programmable frequency level, and a spectral flatness of the frequency response signal is calculated in a selected frequency band surrounding the local minimum. If the spectral flatness is greater than a selected threshold value, acoustic correction is applied to the frequency response in the selected frequency band using an estimated transfer function to obtain a corrected sound pressure frequency response. The corrected sound pressure frequency response is used to modify, or make a recommendation to modify, a physical or operational characteristic of the hearing device.

An audio device comprising an interface with a receiver with an output sound based on an audio output signal; a feed-forward microphone; a distributor for provision of a first feed-forward input signal and a second feed-forward input signal; a feed-back microphone; a feed-forward filter for provision of a feed-forward output signal; an intermediate filter module, wherein the intermediate filter module comprises a hear-through filter based on the second feed-forward input signal and wherein the intermediate filter module is configured to provide the intermediate signal based on the hear-through signal and the audio input signal; a first mixer for provision of an error signal based on the feed-back input signal and the intermediate signal; a feed-back filter signal based on the error signal; and a second mixer for provision of the audio output signal based on the feed-forward output signal and the feed-back output signal.

The present disclosure relates to a method and system for improving the auditory experience of a wearer's own speech by optimizing the speaker output of a PSAP or hearing aid. The method may include: determining whether a user's own speech is present based on a bone conduction signal received by a bone conduction sensor; in response to determining that the user's own speech is present, determining whether ambient sound is present based on an air conduction microphone signal received by the air conduction microphone; and in response to determining that the ambient sound is not present, performing a first adjustment operation on the speaker output based on the bone conduction signal, the air conduction microphone signal, and the transfer function. The method may further include performing a second adjustment operation on the speaker output in response to determining that the ambient sound is present.

A sound signal processing method and a headset device. A target filter and a first audio processing unit are added. The target filter processes an external sound signal collected by an external microphone, to obtain an environmental sound attenuation signal and a voice attenuation signal. The first audio processing unit removes, based on the environmental sound attenuation signal and the voice attenuation signal, a second external environmental sound signal and a second voice signal from an in-ear sound signal collected by an error microphone, to obtain a blocking signal, and transmits the blocking signal to a feedback filter. The feedback filter may generate an inverted noise signal corresponding to the blocking signal and play the inverted noise signal through a speaker.

Secondary path measurements and associated acoustic transducer-to-eardrum responses are obtained from test subjects. Both a least squares estimate and a reduced dimensionality estimate are determined that both estimate a relative transfer function between the secondary path measurements and the associated acoustic transducer-to-eardrum responses. An individual secondary path measurement for a user is performed based on a test signal transmitted via a hearing device into an ear canal of the user. An individual cutoff frequency for the individual secondary path measurement is determined. First and second acoustic transducer-to-eardrum responses below and above the cutoff frequency are determined using the individual secondary path measurement and the least squares estimate. A sound pressure level at an eardrum of the user can be predicted using the first and second receiver-to-eardrum responses.

A method allowing a user to hear own real voice is provided herein. The method may eliminate a part of the user's voice which is conducted through the user's bone vibrations from his / her throat and can be heard by the user. Such bone signal can be superimposed and neutralized by playing a reversed bone signal to the user's ear. The reversed bone signal can be generated with opposite phase and same amplitude and frequency as the bone signal, and played through a signal player to the user's ear. A system allowing a user to hear own real voice is also provided herein.

A hearing aid configured to be worn by a user at or in an ear of the user, comprises • an ITE-part adapted for being located at or in an ear canal of the user; • at least one first input transducer configured to provide corresponding at least one first electric input signal representing sound; • a hearing aid processor configured to provide a processed signal in dependence of said at least one electric input signal; • a first output transducer configured to play sound to the user in dependence of said processed signal, or a signal dependent thereon; • an active occlusion cancellation system (AOCS) for providing an acoustic anti-occlusion signal configured to cancel or diminish a sense of occlusion of the user when the user is speaking, or otherwise is using his or her voice, or when otherwise moving the jaws; wherein the active occlusion cancellation system comprises • an ear canal input transducer located in said ITE-part and configured to provide an electric ear canal input signal representing sound in said ear canal, when the user wears the hearing aid; • an ear canal sound estimation unit (ECSE) configured to estimate sound in said ear canal and to provide an electric anti-occlusion signal in dependence of said electric ear canal input signal and said processed signal; and • a second output transducer located in said ITE-part configured to play sound to the user and to provide said acoustic anti-occlusion signal in dependence of said electric anti-occlusion signal, wherein said second output transducer is specifically adapted to provide sound at frequencies below a threshold frequency. A method of operating a hearing aid is further disclosed.

In a method for actively suppressing occlusion effects when reproducing an audio signal with a headphone (10) or a hearing aid according to the present application, an external sound signal is picked up (20) with at least one external microphone (11) of the headphone or the hearing aid. A speech signal is picked up (21) with at least one additional microphone (12, 17). A dry component of the picked-up speech signal is estimated (22), wherein the dry component of the picked-up speech signal is a part of the picked-up speech signal without reverberation or ambient noise caused by the surrounding space. A speech component is extracted (23) from the external sound picked up with the at least one external microphone by a filter, wherein filter coefficients of the filter are determined (23) based on the estimated dry component of the picked-up speech signal, or the estimated dry component of the picked-up speech signal is filtered such that a speech component is generated (23) having a similar spatiality as the speech component at the external microphone. The extracted or generated speech component is output (24) by a loudspeaker of the headphone or the hearing aid.

Digital audio signal processing techniques used to provide an acoustic transparency function in a pair of headphones. A number of transparency filters can be computed at once, using optimization techniques or using a closed form solution, which are based on multiple re-seatings of the headphones and that are as a result robust for a population of wearers. In another embodiment, a transparency hearing filter of a headphone is computed by an adaptive system that takes into consideration the changing acoustic to electrical path between an earpiece speaker and an interior microphone of that headphone while worn by a user. Other embodiments are also described and claimed.

A hearing aid configured to be worn by a user at or in an ear of the user, comprises • an ITE-part adapted for being located at or in an ear canal of the user; • at least one first input transducer configured to provide corresponding at least one first electric input signal representing sound; • a hearing aid processor configured to provide a processed signal in dependence of said at least one electric input signal; • a first output transducer configured to play sound to the user in dependence of said processed signal, or a signal dependent thereon; • an active occlusion cancellation system (AOCS) for providing an acoustic anti-occlusion signal configured to cancel or diminish a sense of occlusion of the user when the user is speaking, or otherwise is using his or her voice, or when otherwise moving the jaws; wherein the active occlusion cancellation system comprises • an ear canal input transducer located in said ITE-part and configured to provide an electric ear canal input signal representing sound in said ear canal, when the user wears the hearing aid; • an ear canal sound estimation unit (ECSE) configured to estimate sound in said ear canal and to provide an electric anti-occlusion signal in dependence of said electric ear canal input signal and said processed signal; and • a second output transducer located in said ITE-part configured to play sound to the user and to provide said acoustic anti-occlusion signal in dependence of said electric anti-occlusion signal, wherein said second output transducer is specifically adapted to provide sound at frequencies below a threshold frequency. A method of operating a hearing aid is further disclosed.

Earpieces and methods for acute sound detection and reproduction are provided. A method can include measuring an external ambient sound level (xASL), monitoring a change in the xASL for detecting an acute sound, estimating a proximity of the acute sound, and upon detecting the acute sound and its proximity, reproducing the acute sound within an ear canal, where the ear canal is at least partially occluded by an earpiece. Other embodiments are disclosed.

In general, in one aspect, a hearing aid has an ANR circuit and an ear tip that acoustically occludes the ear. Such a hearing aid provides greater gain to sounds than would be stable in the same hearing aid with a vented ear tip. The ear tip and the ANR circuit in combination attenuate sounds reaching the ear canal through the hearing aid to a first level. The hearing aid detects sounds arriving at a microphone, amplifies those sounds, and provides the amplified sounds to the ear canal at a second level and later in time than the same sounds arrive at the ear canal through the ear tip. The first level is at least 14 dB greater than the second level, such that the amplified sounds do not interact with the passive sounds to result in spectral combing.

The present disclosure relates to a method, device, earphone and storage medium for reducing the occlusion effect of earphones. The method for reducing the occlusion effect of earphones is applied to earphones, and the method for reducing the occlusion effect of earphones includes: in response to detecting that a user makes a sound, collecting audio data inside and outside the ear canal, performing spectrum analysis on the audio data, and obtaining a first frequency response curve and a second frequency response curve; determining a first target frequency interval; determining a target frequency response curve based on the first frequency response curve and the second frequency response curve within the first target frequency interval; using the frequency response curve of the target frequency response curve within the second target frequency interval as a compensation curve, and based on the compensation curve, determining multiple target filter parameters, and generating a compensation signal for reducing the occlusion effect of the earphones. Through the present disclosure, the earphones can optimize the effect of reducing the occlusion effect of the earphones.

A hearing device comprises an earpiece adapted to fit the ear canal of a user, so as to separate an inner ear canal volume rom an outside. The hearing device comprises an electroacoustic transducer (11) in acoustic communication with the inner ear canal volume, the electroacoustic transducer (11) being capable of transforming an electrical signal between terminals (16, 17) into an acoustic signal. The hearing device is also equipped to process hearing device input into a wanted signal, to use the wanted signal to obtain a transducer input signal, and to apply the transducer input signal to the terminals (16, 17) of the electroacoustic transducer (11). The hearing device is further equipped to measure a physical signal at the terminals (16, 17) and to use the measured physical signal as another input quantity for obtaining the transducer input signal.

Embodiments of this application provide example signal processing methods, example apparatuses, and example device control methods and apparatuses. One example signal processing method is applied to a hearing aid apparatus. The method includes collecting a first signal and a second signal if it is detected that a user wears the hearing aid apparatus and the user makes a sound, where the first signal includes a self-speaking voice of the user and an ambient sound, and the second signal includes a sound signal of the user.