22results about "Occlusion effect electronic compensation" patented technology

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.

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.

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.

This disclosure provides a speech enhancement method, apparatus, electronic device, storage medium, and program, relating to the field of audio processing technology. The specific steps are as follows: in response to detecting that a user is wearing headphones, a prompt audio data is emitted through a speaker in the headphones, and audio data within the ear canal is collected; a target occlusion effect curve is determined based on the prompt audio data and the audio data within the ear canal; a second sound signal is filtered to obtain a third sound signal; and a noise signal is calculated based on the first sound signal to enhance the third sound signal. This disclosure determines the frequency response curve within the ear canal and the occlusion effect curve by using the prompt audio data and the audio data within the ear canal, thereby enhancing the third sound signal, avoiding high computational resource consumption during speech enhancement, and improving the efficiency of speech enhancement.

In a method according to the invention for processing ambient sounds acquired by an audio device (10) capable of being worn on or in the ear, a reference signal is generated (S1) as a function of the acquired ambient sounds. A measurement signal is generated (S1) from sound acquired in an ear canal of a user of an audio device capable of being worn on or in the ear. The reference signal is processed (S2) into a first signal and the measurement signal is processed (S2) into a second signal. A factor is determined (S3) based on the first signal and the second signal such that the scale for measuring the signal strength of the second signal matches the scale for measuring the signal strength of the first signal. A third signal based on the reference signal is weighted (S4) with the determined factor. And outputting (S5) a third signal weighted with the measured factor.

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 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.

An ear-worn hearing device with occlusion reduction is disclosed. The hearing device includes a signal processor configured to generate an anti-occlusion signal based on a feedforward signal from a vibration sensor located to detect tissue-propagated vibration within the user's at least partially occluded ear canal. Optionally, the anti-occlusion signal can also be based on a feedback signal from a microphone located to detect sound within the at least partially occluded ear canal. Unwanted vibrations detected by the vibration sensor can optionally be removed from the anti-occlusion signal based on filtering the anti-occlusion signal.

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

Embodiments of this application are applicable to the field of electronic technologies, and provide 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. Therefore, the feedback filter does not need to weaken the second external environmental sound signal and the second voice signal in the in-ear sound signal. In this way, not only is a blocking effect suppressed, but a restoration degree of the first external environmental sound signal and the first voice signal sent by a user is improved.

A hearing device comprising a body provided with a dome), wherein a vent path is provided through the body, and wherein the body is further provided with a valve and a valve driver which connects to the valve for opening and closing of the vent path, wherein when the vent path is closed the valve engages the body, and wherein said valve driver comprises a permanent or switchable magnet and a first electrical coil for generating magnetic field lines that cooperate with the permanent of switchable magnet for moving the valve, wherein the valve driver is provided with a second electrical coil for generating magnetic field lines that cooperate with the permanent of switchable magnet, and in that when the valve driver is active for opening of closing of the vent path, the first electrical coil and the second electrical coil are energized individually so as to avoid acceleration of the valve upon engaging the body for opening of closing of the vent path.

An ear-plug assembly presents audio content to an ear canal of a user. The audio content may be based in part on sound in a local area surrounding the user. The ear-plug assembly detects, via one or more acoustic sensors, sound in the area around the user. The sound waves travel through an aperture in a body of the ear-plug assembly and are propagated to a waveguide to the one or more acoustic sensors. The ear-plug assembly processes the detected sound data in a controller, which instructs a speaker assembly to present audio content based in part on the detected sound data. The detected sounds may be amplified, attenuated, filtered, and / or augmented when presented by the speaker assembly.

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.