The invention relates to a variable-directivity MEMS microphone. The microphone comprises an acoustic cavity. The following components are provided inside the acoustic cavity: a first acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a first acoustic conversion signal; a first pre-amplifer, connected to the first acoustic transducer, and configured for outputting a first electric signal; a second acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a second acoustic conversion signal; a second pre-amplifer, connected to the second acoustic transducer, and configured for outputting a second electric signal; and a signal processing chip, connected to the first pre-amplifer and the second pre-amplifer, and configured for generating a directional output signal by performing an arithmetic operation on the first electric signal and the second electric signal under the action of a switching control signal.

A first device includes a memory configured to store instructions and one or more processors configured to receive audio signals from multiple microphones. The one or more processors are configured to process the audio signals to generate direction-of-arrival information corresponding to one or more sources of sound represented in one or more of the audio signals. The one or more processors are also configured to and send, to a second device, data based on the direction-of-arrival information and a class or embedding associated with the direction-of-arrival information.

A speaker system includes a first speaker and a second speaker. The first speaker includes a first sound signal input interface configured to acquire a first sound signal, and a first sound emitter configured to output a first sound based on the first sound signal in a state in which a position of a host device is fixed. The second speaker includes a second sound signal input interface configured to acquire a second sound signal, a moving body that includes a position information acquisition interface configured to acquire information relating to position and that is configured to move based on the information relating to position, and a second sound emitter configured to output a second sound based on the second sound signal.

Techniques described herein include generating first audio signals representative of sounds emanating from an environment and captured with an array of microphones disposed within an ear-mountable listing device. A rotational position of the array of microphones is determined. A rotational correction is applied to the first audio signals to generate a second audio signal. The rotational correction is based at least in part upon the determined rotational position. A speaker of the ear-mountable listening device is driven with the second audio signal to output audio into an ear.

In the acoustic object extraction device, beam forming processing units generate a first acoustic signal by beam forming in an arrival direction of a signal from an acoustic object with respect to a microphone array and generate a second acoustic signal by beam forming in an arrival direction of a signal from the acoustic object with respect to a microphone array, and a common component extraction unit extracts, on the basis of a similarity between the spectrum of the first acoustic signal and the spectrum of the second acoustic signal and from the first acoustic signal and the second acoustic signal, a signal containing a common component corresponding to the acoustic object. The common component extraction unit divides the spectrums of the first acoustic signal and the second acoustic signal into a plurality of frequency sections and calculates a similarity for each of the frequency sections.

A flux beam is generated as a function of flux magnitude patterns. A plurality of flux signals is detected via a sensor array comprising a plurality of sensors. A plurality of flux patterns is generated based on the plurality of flux signals, each of the plurality of flux patterns representing a respective one of the plurality of flux signals. A plurality of flux magnitude patterns is generated based on the plurality of flux patterns, each of the plurality of flux magnitude patterns representing an absolute value of a respective one of the plurality of flux patterns. A flux beam is then generated as a function of the plurality of flux magnitude patterns.

Embodiments relate to an audio system for various audio applications. The audio system registers the locations of one or more sound sources and selects the target sound source based on a hidden Markov model. A health monitoring system that integrates an audio system may use information collected by sensors to monitor an amount of social interaction of a user and predict a risk of dementia and/or hearing loss based on a model. The audio system uses a current/voltage sensor to detect electrical drive signals for determining a level of audio leakage of the audio system. Additionally, the audio system may update a video stream with an audio background based on an artificial visual background in the video stream so that the updated video stream sounds as if it originated from the user being located in a physical representation related to the background.

Briefly, in accordance with one or more embodiments, a display includes a housing comprising a first surface and a second surface opposite to the first surface. The second surface comprises a transparent material covering the second surface and the housing includes two or more microphone ports disposed along a parting line between the first surface and the second surface exterior to the transparent material. The housing further includes two or more microphones coupled with the two or more microphone ports. A microphone signal processing system may be utilize to increase directional sensitivity of the two more microphones toward an audio source. An angle detector to detect an angle of the may be utilized to accommodate the directional sensitivity provided by the microphone signal processing system.

Techniques for improving beamforming using filter coefficient values corresponding to virtual microphones are described. A system may define “virtual” microphone positions and determine corresponding filter coefficient values. These filter coefficient values may be applied to input audio data captured by actual physical microphones, enabling the system to improve performance of beamforming and/or to reduce a number of physical microphones without degrading performance. Offline testing and simulations may be performed to identify the best combination of virtual microphones and/or filter coefficient values for a particular look-direction. For example, the simulations may identify that a first filter coefficient corresponding to a first virtual microphone and a first direction will be associated with a first physical microphone and the first direction. During run-time processing, a device may generate beamformed audio data for the first direction by applying the first filter coefficient to input audio data captured by the first physical microphone.

An example method of operation may include detecting an acoustic stimulus via active beams associated with at least one microphone disposed in a defined space, detecting loudspeaker characteristic information of at least one loudspeaker providing the acoustic stimulus, transmitting acoustic stimulus information based on the acoustic stimulus to a central controller, and modifying, via a central controller, at least one control function associated with the at least one microphone and the at least one loudspeaker to minimize acoustic feedback produced by the loudspeaker.