Abstract: A method of configuring lighting fixture microphones includes receiving, by a processor circuit, an electrical audio signal from a first microphone. The method further includes determining, by the processor circuit, whether the first microphone is defective based on the electrical audio signal. The method also includes enabling, by the processor circuit, a second microphone in response to determining that the first microphone is defective.

Abstract: A touch Bluetooth headset includes a housing, a touchpad, a main board, a Bluetooth antenna, a first thimble and a second thimble. The touchpad is a flexible touchpad, the main board and the touchpad are opposite and fixed on the housing, and the touchpad is provided with a plurality of metal contacts. The first thimble and the second thimble have one end fixed to the main board and other end extending toward the touchpad and abutting against the metal contacts. The Bluetooth antenna is laid on the touchpad and connected to the metal contacts. The touch Bluetooth headset may be designed to be more miniaturized; additionally, an additional Bluetooth antenna is added in the touchpad and the Bluetooth antenna is jacked up by the above thimbles, which reduces the electromagnetic shielding, thereby the micro Bluetooth headset has a better signal transmission distance.

Abstract: The present disclosure provides a speaker box including a shell, a speaker box, a support wall formed by the inner extension of the shell, and a sound guide channel formed in the receiving space. The speaker is fixed to the support wall. The speaker box includes a surrounding wall formed by the inner extension of the shell and a cover plate arranged on the surrounding wall. The shell, surrounding wall, support wall and cover plate are jointly enclosed as auxiliary sound cavity. The support wall is provided with a through-hole, and the auxiliary sound cavity is communicated with the front cavity through the through-hole to form the resonant cavity structure. The speaker box further includes a leakage to achieve the sound pressure balance inside and outside of the shell. Compared with the related art, the high frequency acoustic performance of the speaker box of the present disclosure is excellent.

Abstract: A wireless earpiece includes a wireless earpiece housing, a processor disposed within the wireless earpiece housing, at least one microphone operatively connected to the processor, and at least one speaker operatively connected to the processor. The processor is configured to receive audio from the at least one microphone, perform processing of the audio to provide processed audio, and output the processed audio to the at least one speaker. The processing of the audio involves identifying body generated sounds generated by a body of a user of the wireless earpiece and removing the body generated sounds.

Abstract: Technology described in this document can be embodied in an earpiece of an active noise reduction (ANR) device. The earpiece includes a plurality of microphones, wherein each of the plurality of microphones is usable for capturing ambient audio to generate input signals for both an ANR mode of operation and a hear-through mode of operation of the ANR device. The earpiece further includes a controller configured to: process a first subset of microphones from the plurality of microphones to generate input signals for the ANR mode of operation, process a second subset of microphones from the plurality of microphones to generate input signals for the hear-through mode of operation, detect that a particular microphone of the second subset is acoustically coupled to an acoustic transducer of the ANR device in the hear-through mode of operation, and in response to the detection, process the input signals from the second subset of microphones without using input signals from the particular microphone.

Abstract: A system configured to detect microphone failure and optimize settings using remaining functional microphones. For example, a device may detect a defective (e.g., nonfunctional or malfunctioning) microphone when an energy level for the defective microphone is lower than a threshold value for a period of time. After detecting the defective microphone, the device may update configuration settings by reassigning microphones (e.g., selecting a functional microphone instead of the defective microphone) and/or by calculating new configuration settings accordingly (e.g., excluding a nonfunctional microphone or compensating for an attenuation or phase shift of a malfunctioning microphone). For example, the system may recalculate beamformer coefficients using the remaining microphones, which may improve a performance of the device (e.g., wakeword detection, automatic speech recognition (ASR), etc.) and result in only a marginal performance degradation relative to fully-functioning microphones.

Abstract: A system identification device for performing fast real-time identification for a system from input/output data includes a filter robust to disturbance, by setting the maximum energy gain from the disturbance to a filter error, as an evaluation criterion, smaller than a given upper limit. The filter estimates a state estimation value of a state of the system.

Abstract: Audio (noise source) is output as a cancellation sound through a variable filter and a first filter, and transmitted to the second filter. A subtractor subtracts an output of a second filter from an output of a microphone, and an adaptive algorithm execution unit updates a transfer function of the variable filter so that the subtracted result becomes zero (0). A transfer function A for the first filter is a transfer function which can cancel noise at a position of a user's ear by setting, as the cancellation sound, a sound obtained by applying the transfer function A to audio at the time of learning, and a transfer function B for the second filter is a transfer function which can eliminate, for the cancellation sound, a difference between a sound obtained by applying the transfer function B to audio and the output of the microphone.

Abstract: A method and an apparatus for locating a sound source are provided. The method includes: obtaining M channels of audio signals of a preset format by microphone arrays located in different planes (S100); preprocessing the M channels of audio signals of the preset format, and projecting them onto the same plane, so as to obtain N channels of audio signals, where M?N (S200); performing a time-frequency transformation on each of the N channels of audio signals, so as to obtain frequency domain signals of the N channels of audio signals (S300); further calculating a covariance matrix of the frequency domain signals and performing a smoothing process (S400); performing an eigenvalue decomposition of the smoothed covariance matrix (S500); estimating the sound source direction according to an eigenvector corresponding to the maximum eigenvalue, so as to obtain a sound source orientation parameter (S600).

Abstract: One or more embodiments of the present disclosure utilize two distinctly different radiation devices aimed in the same direction to create a derived acoustic radiation pattern. The radiation devices may be distinctly different in terms of the acoustic radiation pattern each radiation device generates individually. The derived acoustic radiation pattern is unique to the individual acoustic radiation patterns of either of the two radiation devices. Manipulation of several key design variables allows a multitude of unique patterns to be derived in this way using only the two radiation devices. In turn, this allows for engineering an acoustic radiation pattern to match the unique geometry of a room.

Abstract: Voice responsive in-wall devices are provided. In one example implementation, a power switch includes a housing mountable on or at least partially within a surface. The housing can have a front panel. The power switch can include an interface element disposed on the front panel and operable to receive a user input. The power switch can include a power interrupter operable to control power delivery to the powered load based at least in part on interaction with the interface element. The power switch can include one or more microphones operable to obtain audio input. The power switch can include one or more speakers configured to provide audio output. The power switch can include a communications interface operable to communicate data associated with the audio input over a communication link.

Abstract: The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

Abstract: An active noise reduction (ANR) earphone with a housing comprising a front cavity and an exit that is fluidly coupled to the front cavity. An acoustic driver is configured to deliver acoustic energy into the front cavity of the housing. A nozzle is coupled to the housing and is configured to direct acoustic energy from the housing exit to a nozzle exit opening. A microphone is located in the nozzle. A compliant sealing structure is coupled to the nozzle and is configured to couple the earphone to an ear of a user.

Abstract: A method of estimating an individualized head-related transfer function and an individualized interaural time difference function of a particular person, comprises the steps of: a) obtaining a plurality of data sets comprising a left and a right audio sample from in-ear microphones, and orientation information from an orientation unit, measured in a test-arrangement where an acoustic test signal is rendered via a loudspeaker and the person is moving the head; b) extracting interaural time difference values and/or spectral values, and corresponding orientation values; c) estimating a direction of the loudspeaker relative to the head using a predefined quality criterion; d) estimating an orientation of the orientation unit relative to the head; e) estimating the individualized ITDF and the individualized HRTF. A computer program product may be provided for performing the method, and a data carrier may contain the computer program.

Abstract: A method expedites playing sound of a talking emoji from a first person with a first portable electronic device (PED) to a second person with a second PED. The second PED receives the talking emoji in mono sound and convolves the mono sound into binaural sound before receiving a request to play the sound to the second user. The second PED then plays the sound of the talking emoji in binaural sound after receiving the request from the second user.

Abstract: An example active noise control filtering with an adaptive filter structure includes a controllable filter matrix with reference and error input signals, and updating the filter coefficients dependent on an optional filtered reference signal and an error signal, the error signal being representative of a performance criterion of the filter module. Further, a leakage functionality and a convergence functionality is applied to the updated filter coefficients. The leakage functionality is controlled by at least one of a flush functionality, freeze functionality, spatial freeze functionality and leakage threshold, and the convergence functionality is controlled by the freeze functionality and spatial freeze functionality.

Abstract: Audio information of audio content being listened to by a user is received. An aspect of a listening environment of the user is identified. A volume preset, based on the audio information and the aspect of the listening environment, is determined to be available. A first volume of the audio content being listened to by the user is determined to be different from the volume preset. The first volume of the audio content is adjusted to a second volume.

Abstract: The present application is applicable to the technical field of intelligent home, and provides a method and a device for switching play modes of a wireless speaker, and a wireless speaker. The method for switching play modes of a wireless speaker includes: acquiring user information in a current application scenario by the camera and/or the microphone array; and controlling the wireless speaker to switch into a play mode corresponding to the user information. The embodiments of the present application can automatically switch the play modes according to the application scenario, thereby improving the switching efficiency, reducing the waiting time of the user, and having strong ease of use and practicability.

Abstract: Embodiments are described for a method for localizing a set of speakers (106) and microphones (108), having only the times of arrival between each of the speakers and microphones. An autodiscovery process (107) uses an external input to set: a global translation (3 continuous parameters), a global rotation (3 continuous parameters), and discrete symmetries, i.e., an exchange of any axis pairs and/or reversal of any axis. Different time of arrival acquisition techniques may be used, such as ultrasonic sweeps or generic multitrack audio content. The autodiscovery algorithm is based in minimizing a certain cost function, and the process allows for latencies in the recordings, possibly linked to the latencies in the emission.

Abstract: A sound bar is provided which includes a generally elongate rectangular structure, with one angled output face having a plurality of speakers disposed therein, to thereby provide enhanced audio output. The sound bar is beneficially provided with a single mounting surface configured and arranged such that the sound bar can be mounted on both a table/shelf (horizontal orientation) and a wall (vertical orientation) via the single mounting surface by rotating the sound bar 180°, wherein the audio output in both mounting orientations is a multi-directional 3D audio output that travels to and bounces off both the walls and the ceiling of a room in which the sound bar is located.