Abstract: An earpiece including an intra-auricular support positionable in an auricle of an ear, a portion of the intra-auricular support configured as an intra-auricular support base securable to an earphone housing of an earphone.

Abstract: Provided a robot for assisting hearing of a user, while minimizing an influence on the surroundings. The robot includes a speaker, a microphone configured to recognize a voice, a processor configured to acquire a position of a user's face when a hearing aid command is acquired on the basis of the voice recognized through the microphone, and a driving unit configured to cause the speaker to be moved toward the position of the user's face, wherein the processor acquires a sound which is a target of hearing aid, generates an assistant sound by amplifying a predetermined frequency band of the sound or converting the predetermined frequency band of the sound into a different frequency band, and outputs the assistant sound through the speaker.

Abstract: Exemplary noise reducing methods and systems are configured to acoustically reproduce two electrical noise reducing signals at two opposing positions in a helmet; to pick up sound at positions in the vicinity of the positions where the noise reducing signals are reproduced; and to generate the two noise reducing signals from error-signals and reference-signals. For each noise reducing signal, the corresponding error-signal is generated from the sound picked-up at the same position where the respective noise reducing signal is reproduced; and the reference-signal is generated from the sound picked-up at the position where the respective other noise reducing signal is reproduced.

Abstract: Systems, methods, and apparatus are provided for recording high output power levels of sound at low sound pressure levels. For example, an apparatus comprises an enclosure, a speaker disposed within the enclosure, a microphone disposed within the enclosure, and an evacuation port. The evacuation port is configured to connect to a system that reduces a pressure level within the enclosure to a level that is less than an ambient air pressure level outside the enclosure. The enclosure is sealed or otherwise configured to provide a sealed enclosure, to maintain the reduced air pressure within the enclosure. The speaker can be driven by an amplifier at high output power levels to generate a distorted sound of an amplified electric musical instrument for recording purposes, while the reduced pressure level within the enclosure serves to attenuate the sound pressure level and perceived loudness which emanates from the speaker.

Abstract: A media system and a method of using the media system to accommodate hearing loss of a user, are described. The method includes selecting a personal level-and-frequency dependent audio filter that corresponds to a hearing loss profile of the user. The personal level-and-frequency dependent audio filter can be one of several level-and-frequency-dependent audio filters having respective average gain levels and respective gain contours. An accommodative audio output signal can be generated by applying the personal level-and-frequency dependent audio filter to an audio input signal to enhance the audio input signal based on an input level and an input frequency of the audio input signal. The audio output signal can be played by an audio output device to deliver speech or music that the user perceives clearly, despite the hearing loss of the user. Other aspects are also described and claimed.

Abstract: A first digital signal processor (DSP) associated with a first user outputs audio to the first user via first headphones, and captures speech from the first user via a first microphone. Similarly, a second DSP associated with a second user outputs audio to the second user via second headphones, and captures speech from the second user via a second microphone. The first DSP is coupled to the second DSP in order to allow the first and second users to share music and communicate with one another. The first user may speak into the first microphone, and the first and second DSPs may then interoperate to output that speech to the second user without substantially disrupting audio output to the second user. Each of the first and second users may also select between first and second audio sources that may be coupled to the first and second DSPs, respectively.

Abstract: In an example implementation, a self-cooling headset includes an ear cup to form an ear enclosure volume and a control volume. The headset also includes an intake valve to open and admit air from the ear enclosure volume into the control volume when a negative pressure is generated within the control volume, and an exhaust valve to open and release air from the control volume into the ambient environment when a positive pressure is generated within the control volume.

Abstract: Adaptive filters output a cancellation sound from a speaker, a selector selects outputs of a plurality of auxiliary filters each corresponding to different positions, a subtractor subtracts the selected output from the output of the microphone and outputs the subtracted output to the adaptive filter as an error signal, and a position detection device detects a position of a head of a user. A transfer function estimated so that the error signal becomes 0 when noise is canceled at the corresponding position is preset in the auxiliary filter. When the auxiliary filter corresponding to the position close to the head of the user changes, the switching control unit stepwise increases the frequency with which the output of the auxiliary filter is selected by the selector to 100%.

Abstract: Disclosed are hearing protection devices and housings for noise sensors for the same. Hearing protection devices can include an ear cup including an external casing partially defining an inner space, a noise sensor including a microphone electrically coupled to a printed circuit board, and a housing disposed in an aperture defined in the external casing. The housing can define an axial bore defining a noise sensor receiving portion and an acoustic communication portion. The inner space of the ear cup can be substantially airtight when the housing is sealably disposed at or proximate the aperture, the microphone is engaged within the noise sensor receiving portion of the housing, and the ear cup is worn securely about the wearer's ear. The noise sensor can be calibrated by removing a removable securing collar and slidably disposing a calibration tool into the axial bore without further disassembling the hearing protection device.

Abstract: The disclosure is generally directed to a system for reducing wind noise. A system includes one or more processors coupled to a non-transitory computer-readable storage medium having instructions encoded thereon that, when executed by the one or more processors, cause the one or more processors to obtain signals respectively generated from two or more microphones during a time period, the signals representing acoustic energy detected by the two or more microphones during the time period, determine a coherence between the signals, and determine a filter based on the coherence. The filter is configured to reduce wind noise in one or more of the signals.

Abstract: In at least one embodiment, a microphone apparatus for attachment to an exterior portion of a vehicle is provided. The apparatus includes a printed circuit board (PCB), a microphone element, a cover, and a grille cover. The microphone element is positioned on the PCB. The cover extends over the printed circuit board and defines a first opening to enable ambient sound external to the vehicle to be transmitted to the microphone element. The grille cover is positioned on the exterior portion of the vehicle defining a plurality of openings for directly receiving the ambient sound and for transmitting the ambient sound to the first opening of the cover and to the microphone element.

Abstract: Techniques are described for reducing sensitivity to non-acoustic stimuli. In some embodiments, differential beamforming is applied to microphone signals generated based on responses of microphones to an acoustic stimulus and a non-acoustic stimulus. Compensated signals can be generated based on the microphone signals such that the compensated signals are in phase with respect to the acoustic stimulus. The non-acoustic stimulus is detectable by comparing a first signal to a second signal to determine that one signal has a greater instantaneous magnitude. The first signal can be a beamformed signal or signal derived therefrom, and the second signal can be an average of the compensated signals or signal derived therefrom. An output audio signal can be generated by switching or cross fading between the beamformed signal and a noise-reduced signal such that a contribution of the noise-reduced signal is increased and a contribution of the beamformed signal is decreased.

Abstract: A public address method for live broadcast, in a helmet, of an audio signal conditioned from a plurality of raw audio channels, includes a pre-processing phase including the operations that consist of taking into account characteristics of the auditory perception of the listener; correcting each channel as a function of the characteristics of the auditory perception of the listener; a mixing phase including the production, from the channels thus pre-processed, of a mixed audio signal; a post-processing phase including the operations that consist of: measuring the sound level of a background noise; correcting the mixed audio signal as a function of the sound level of the background noise; a phase of reproducing, in the helmet, the conditioned audio signal resulting from post-processing.

Abstract: A noise cancellation filter structure for a noise cancellation enabled audio device, in particular headphone, comprises a noise input for receiving a noise signal and a filter output for providing a filter output signal. A first noise filter produces a first filter signal by filtering the noise signal and a second noise filter produces a second filter signal by filtering the noise signal. The second noise filter has a frequency response with a non-minimum-phase, in particular maximum-phase. A combiner is configured to provide the filter output signal based on a linear combination of the first filter signal and the second filter signal.

Abstract: A media system and a method of using the media system to accommodate hearing loss of a user, are described. The method includes selecting a personal level-and-frequency dependent audio filter that corresponds to a hearing loss profile of the user. The personal level-and-frequency dependent audio filter can be one of several level-and-frequency-dependent audio filters having respective average gain levels and respective gain contours. An accommodative audio output signal can be generated by applying the personal level-and-frequency dependent audio filter to an audio input signal to enhance the audio input signal based on an input level and an input frequency of the audio input signal. The audio output signal can be played by an audio output device to deliver speech or music that the user perceives clearly, despite the hearing loss of the user. Other aspects are also described and claimed.

Abstract: Audio device and method for designing and making a diaphragm, the audio device comprising a diaphragm having a curved profile adapted for radiation of audio signals from a plurality of bending modes and a piston mode, one or more of the plurality of bending modes having coincident nodal line locations, the diaphragm having a frontal side and a rear side, and a transducer coupled to the rear side of the diaphragm, the transducer adapted for driving the diaphragm for radiation of audio signals having reduced audio distortion, wherein the plurality of bending modes each have minima locations throughout the diaphragm, and wherein the transducer is mounted on one of the minima locations of the plurality of bending modes and one or more impedance components are mounted on at least one of the remaining minima locations to inertially balance the diaphragm based on a pre-determined relative mean modal velocity limit.

Abstract: Methods, systems, and computer program products that infer and correct automatic gain compensation (AGC) values over time are described. A device emits a series of inaudible reference audio signals during recording. The reference audio signals have a constant amplitude. A microphone of the device records the reference audio signals while recording audio content. The device may apply AGC during the recording. An AGC inference module receives the recorded signals and extracts a representation of the reference audio signals, which have been subject to the AGC. The AGC inference module determines variations in levels of the extracted representation of the reference audio signals over time. Based on the variations, the AGC inference module infers the AGC applied to the recording over time. The AGC inference module can then provide the AGC for reading, or undo effects of the AGC applied to the audio content.

Abstract: Provided are an audio signal processing method and device. The method can include acquiring audio signals from at least two sound sources through at least two microphones to obtain multiple frames of original noise signals of the at least two microphones in a time domain, and, for each frame in the time domain, acquiring respective frequency-domain estimated signals of the at least two sound sources according to the respective original noise signals. The method can further include, for each sound source, dividing the frequency-domain estimated signal into frequency-domain estimated components which each corresponds to a frequency-domain sub-band and includes multiple frequency point data in a frequency domain, determining a weighting coefficient of each frequency point in the frequency-domain sub-band, and updating a separation matrix of each frequency point according to the weighting coefficient and obtaining the audio signals based on the updated separation matrices and the original noise signals.

Abstract: The present invention provides for an apparatus, system, and method for generating a head related audio transfer function in real time. Specifically, the present invention utilizes unique structural components including a tragus structure and an antihelix structure in connection with a microphone in order to communicate the location of a sound in three-dimensional space to a user. The invention also utilizes an audio processor to digitally process the head related audio transfer function.

Abstract: Methods and apparatus for predictive head-tracked binaural audio rendering in which a rendering device renders multiple audio streams for different possible head locations based on head tracking data received from a headset, for example audio streams for the last known location and one or more predicted or possible locations, and transmits the multiple audio streams to the headset. The headset then selects and plays one of the audio streams that is closest to the actual head location based on current head tracking data. If none of the audio streams closely match the actual head location, two closest audio streams may be mixed. Transmitting multiple audio streams to the headset and selecting or mixing an audio stream on the headset may mitigate or eliminate perceived head tracking latency.