Abstract: In one embodiment, an article of manufacture has microphones mounted at different locations on a non-spheroidal device body and a signal-processing system that processes the microphone signals to generate a B Format audio output having a zeroth-order beampattern signal and three first-order beampattern signals in three orthogonal directions. The signal-processing system generates at least one of the first-order beampattern signals based on effects of the device body on an incoming acoustic signal. The microphone signals used to generate each first-order beampattern signal have an inter-microphone effective distance that is less than a wavelength at a specified high-frequency value (e.g., <4 cm for 8 kHz). In preferred embodiments, the inter-microphone effective distance is less than one-half of that wavelength (e.g., <2 cm for 8 kHz).

Abstract: Systems, methods, and devices for monitoring industrial equipment using audio are described herein. One system includes two computing devices. The first computing device can receive, from an audio sensor, audio sensed during operation of industrial equipment, extract a plurality of features from the audio, determine whether any portion of the audio is anomalous, and send, upon determining a portion of the audio is anomalous, the anomalous portion of the audio to the second, remotely located, computing device. The second computing device can provide the anomalous portion of the audio to a user to determine whether the anomalous portion of the audio corresponds to a fault occurring in the equipment, and receive, from the user upon determining the anomalous portion of the audio corresponds to a fault occurring in the equipment, input indicating the anomalous portion of the audio corresponds to the fault to learn fault patterns in the equipment.

Abstract: One embodiment provides a device comprising a speaker driver, a microphone configured to obtain a measurement of a near-field sound pressure of the speaker driver, and a controller. The controller is configured to determine a velocity of a diaphragm of the speaker driver, and automatically calibrate sound power levels of audio reproduced by the speaker driver based on the velocity and the measurement of the near-field sound pressure to automatically adjust the sound power levels to an acoustic environment of the device.

Abstract: A system, method, and wireless earpieces for managing power settings. Sensor measurements are performed utilizing a first sensor array of the wireless earpieces to detect light and motion. Sensor measurements are performed utilizing a second sensor array of the wireless earpieces to detect light and motion. The sensor measurements are analyzed from the first sensor array and the second sensor array. A determination is made whether a change event is detected in response to the sensor measurements. The change event is confirmed as detected. The wireless earpieces enter a full power mode in response to the change event being confirmed.

Abstract: Methods, systems, and media for voice communication are provided. In some embodiments, a system for voice communication is provided, the system including: a first audio sensor that captures an acoustic input; and generates a first audio signal based on the acoustic input, wherein the first audio sensor is positioned between a first surface and a second surface of a textile structure. In some embodiments, the first audio sensor is positioned in a region located between the first surface and the second surface of the textile structure. In some embodiments, the first audio sensor is positioned in a passage located between the first surface and the second surface of the textile structure.

Abstract: The present disclosure provides an electronic device and methods for operating the electronic device. The electronic device may include: a housing having a coupling member removably attachable to an ear of a user; one or more microphones provided within the housing and configured to detect an external sound; at least one speaker provided within the housing; at least one communication circuit within the housing; a processor provided within the housing and electrically coupled to the one or more microphones, the at least one speaker, and the at least one communication circuit; and at least one memory provided within the housing, and electrically coupled to the processor.

Abstract: Embodiments of the present invention disclose a method, computer program product, and system for controlling the volume of microphones in an online meeting. The computer may receive a plurality of voice data from a plurality of user microphones. One or more users in a same room may be identified using the plurality of received voice data and a voice picker, wherein the one or more users in the same room have a same watermark in a set of voice data. A main voice of a user may be recognized from the one or more users in the same room using the voice picker. A microphone volume may be adjusted for each of the one or more users in the same room, wherein the microphone volume of the main voice of the user is increased.

Abstract: A motor vehicle sound generator system has a housing in which a loudspeaker is accommodated. The housing has at least a first orifice and a second orifice. The orifices each have a sound transmission member provided thereon. The first orifice is arranged in a front side region of the loudspeaker and the second orifice is arranged in a rear side region of the loudspeaker as viewed in an axial direction of the loudspeaker. The sound generator system has a first sound outlet opening that is associated with the first orifice and a second sound outlet opening that is associated with the second orifice. The sound exits into open air through the first and second sound outlet openings, with the first and second sound outlet openings being free of exhaust gas flow.

Abstract: The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor.

Abstract: Larger regions are often acoustically irradiated with a plurality of loudspeakers. A public address system 1 for the sonication of a sonication region 2 is proposed, wherein the public address system has a multiplicity of loudspeakers 3a, 3b, wherein the loudspeakers 3a, 3b are arranged in the sonication region 2 are designed to output an audio signal based on an input signal 4a, 4b. The public address system has a measuring device 5 for the detection of the audio signal at a measurement point 6 in the sonication region 2, wherein the measuring device 5 is designed to determine a propagation time difference ?tn of the audio signals to the measurement point 6 between each of two speakers 3a,b and to provide the propagation time difference ?tn as propagation time difference data.

Abstract: An electronic device includes one or more microphones, one or more sensors, and one or more processors operable with the one or more microphones and the one or more sensors. The one or more processors, upon the one or more sensors detecting the electronic device is disposed within a repository container, such as a pocket, apply an audio signal adjustment function to signals received from the one or more microphones, thereby mitigating noise in the signals caused by the repository container.

Abstract: An acoustic processing apparatus includes a detection unit configured to detect a change in a state of a microphone, and a determination unit configured to determine a parameter to be used in acoustic signal generation by a generation unit configured to generate an acoustic signal based on one or more of a plurality of channels of sound collection signals acquired based on sound collection by a plurality of microphones, wherein in a case where a change in at least any of states of the plurality of microphones is detected by the detection unit, the determination unit determines the parameter based on the states of the plurality of microphones after the change.

Abstract: A method for operating a hearing device includes generating first and second input signals from a sound signal by using respective first and second input transducers, providing a first angle and an angular range and, with respect to frequency bands, based on the first and second input signals and first angle, forming an attenuation directional signal having relative attenuation at least for a second angle in the angular range about the first angle and thereby setting and an overlay parameter. A gain directional signal is formed based on the first and second input signals and the overlay parameter and/or second angle, having relative gain for the second angle. An angled directional signal is generated from the attenuation directional signal and the gain directional signal. An output signal is generated based on the angled directional signal. A hearing device and a binaural hearing device system are also provided.

Abstract: A system, method and one or more wireless earpieces for performing self-configuration. A user is identified utilizing the one or more wireless earpieces. Noises from an environment of the user are received utilizing the one or more wireless earpieces. An audio profile associated with the noises of the environment of the user is determined. The components of the one or more wireless earpieces are automatically configured in response to the audio profile and the user identified as utilizing the one or more wireless earpieces.

Abstract: A system and method for reducing or eliminating undesirable acoustic artifacts when a vehicle is struck by road debris. The method includes generating a noise signal representative of a first acceleration detected by an accelerometer of a vehicle caused by a disturbance and generating a noise-cancellation signal via a controller within the vehicle. Residual noise resulting from the combination of the acoustic energy of the noise-cancellation signal and the disturbance is detected by a reference sensor, which generates a reference sensor signal based on the residual noise. The reference sensor signal is transmitted to an adaptive processing module to adapt filter coefficients. A second acceleration is detected by the accelerometer and a level detector calculates the absolute value of a derivative of the second acceleration. If the absolute value exceeds a threshold value, adjustment of the filter coefficients is prevented.

Abstract: In the present invention, a digital electroacoustic transducer apparatus with a noise canceling system includes: a signal processing circuit that generates a digital processing signal based on a digital signal from a sound source; a first voice coil that receives the digital processing signal; a microphone that picks up noise and generates a noise signal; a noise canceling circuit that generates a cancel signal based on the noise signal; and a second voice coil that receives the cancel signal, the first voice coil and the second voice coil driving a diaphragm, thereby avoiding a difference between the phase of the cancel signal and the phase opposite to that of noise.

Abstract: Brownout management for an audio amplification system. An audio amplification system includes audio volume control circuitry, audio sample interpolation circuitry, and brownout management circuitry. The brownout management circuitry includes brownout detection circuitry and brownout response circuitry. The brownout detection circuitry is configured to determine whether a voltage of a battery that powers the audio amplification system is below a brownout threshold, and to generate a brownout detection signal that indicates the voltage is below the brownout threshold. The brownout response circuitry is coupled to an audio output of the audio sample interpolation circuitry. The brownout response circuitry is configured to attenuate the audio samples output by the audio sample interpolation circuitry responsive to the brownout detection signal indicating that the voltage is below the brownout threshold.

Abstract: Embodiments described herein involve a network media system identifying a primary zone in the network media system to play back audio tracks of videos. The system determines respective audio responses of one or more auxiliary zones and identifies a particular auxiliary zone for coordinated playback of auxiliary audio content based on the respective audio responses of the one or more auxiliary zones indicating that the particular auxiliary zone is within audible range of the primary zone and the particular auxiliary zone including a subwoofer. While an audio track of a given video is playing back, the network media system temporarily forms a playback configuration in which the primary zone plays back primary audio content including full frequency range audio content of the audio track and the at least one auxiliary zone plays back auxiliary audio content including low frequency range audio content of the audio track.

Abstract: According to one embodiment, a noise reduction apparatus includes a first control sound source, a first current detection unit, a second control sound source, a second current detection unit, and an adjustment unit. The first control sound source generates a first control sound for reducing noise from a noise source. The first current detection unit detects a first current flowing from the first control sound source upon receiving the noise from the noise source. The second control sound source is provided at a position different from a position of the first control sound source and generates a second control sound for reducing the noise from the noise source. The second current detection unit detects a second current flowing from the second control sound source upon receiving the noise from the noise source. The adjustment unit adjusts the first control sound and the second control sound so as to make the first current and the second current satisfy a predetermined condition.

Abstract: A speech reproduction device for reproducing speech based on a received speech signal so that the reproduced speech is intelligible in a clear speech zone and unintelligible in a masked speech zone includes an audio processing module configured for receiving the speech signal; a set of speech loudspeakers configured for reproducing the speech based on one or more speech loudspeaker signals; and a set of masking sound loudspeakers configured for producing a masking sound based on one or more masking sound loudspeaker signals, wherein the masking sound masks the speech in the masked speech zone; wherein the audio processing module includes a speech signal analysis module configured for producing one or more analysis signals based on spectral and/or temporal characteristics of the speech signal; wherein the audio processing module includes a masking sound generator configured for producing one or more masking sound signals based on the one or more analysis signals.