This disclosure describes a ceiling tile microphone system that includes a plurality of microphones coupled together as a microphone array and used for beamforming processing, one or more separate processing devices that couple to the microphone array, where one or more separate processing devices further include beamforming, acoustic echo cancellation, and adaptive acoustic processing; a single ceiling tile with an outer surface on the front side of the ceiling tile where the outer surface is acoustically transparent, the microphone array combines with the ceiling tile as a single unit, the ceiling tile being mountable in a drop ceiling in place of a ceiling tile included in the drop ceiling; where the system is used in a drop ceiling mounting configuration; where the microphone array couples to the back side of the ceiling tile and all or part of the system is in the drop space of the drop ceiling.

An audio control device and a method of controlling a multi-channel sound system are provided. The audio control device is configured to control the audio output of a multimedia device which includes a first and a second sound receiving devices. The audio control device includes an audio output port and a control circuit which performs the following steps: (A) controlling a test signal to be outputted from the audio output port at a first time point; (B) receiving a first input signal generated by the first sound receiving device, and recording a second time point; (C) receiving a second input signal generated by the second sound receiving device, and recording a third time point; and (D) calculating a speaker position according to the first time point, the second time point, the third time point, and a distance between the first sound receiving device and the second sound receiving device.

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

An allophone inspection device and inspection method thereof are provided. An allophone inspection device includes an array microphone unit in which a plurality of array microphones are disposed at predetermined intervals, and a controller configured to build reference data by quantifying analyzed allophone by collecting sound signals generated from surrounding based on a position where the array microphone unit is installed in advance and measure a surrounding sound signal through the array microphone unit to estimate whether or not noise is generated and a position of the sound source where the noise is generated based on the reference data.

Embodiments include a planar microphone array comprising a first linear array arranged along a first axis; and a second linear array arranged along a second axis orthogonal to the first axis, a center of the second linear array aligned with a center of the first linear array, wherein each of the first linear array and the second linear array comprises a corresponding first set of microphone elements nested within a corresponding second set of microphone elements, and each set of microphone elements is arranged symmetrically about the center of the corresponding linear array, such that the first linear array and the second linear array are configured to generate a steerable directional polar pattern, the microphone elements of each linear array configured to capture audio signals. Embodiments also include a microphone system comprising the same and a method performed by processor(s) to generate an output signal for the same.

A sound data processing device includes: a sound data acquisition unit configured to acquire first sound data that is data about a sound whose sound image is localized in a cabin of a vehicle; an object specifying unit configured to specify an attention object that is an object to which an occupant of the vehicle directs attention; a sound data processing unit configured to generate second sound data that is data about the sound for which a sound relating to the attention object is emphasized in comparison with the first sound data; and a sound data output unit configured to output the second sound data to an output device that outputs a sound to the occupant.

The present disclosure relates to a pair of glasses. The pair of glasses may include a frame, one or more lenses, and one or more temples. The pair of glasses may further include at least one low-frequency acoustic driver, at least one high-frequency acoustic driver, and a controller. The at least one low-frequency acoustic driver may be configured to output sounds from at least two first guiding holes. The at least one high-frequency acoustic driver may be configured to output sounds from at least two second guiding holes. The controller may be configured to direct the low-frequency acoustic driver to output the sounds in a first frequency range and direct the high-frequency acoustic driver to output the sounds in a second frequency range. The second frequency range may include one or more frequencies higher than one or more frequencies in the first frequency range.

A method and a device encode N audio signals, from N microphones where N≥3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

Aspects of a multi-orientation playback device including at least one microphone array are discussed. A method may include determining an orientation of the playback device which includes at least one microphone array and determining at least one microphone training response for the playback device from a plurality of microphone training responses based on the orientation of the playback device. The at least one microphone array can detect a sound input, and the location information of a source of the sound input can be determined based on the at least one microphone training response and the detected sound input. Based on the location information of the source, the directional focus of the at least one microphone array can be adjusted, and the sound input can be captured based on the adjusted directional focus.

An electronic apparatus includes a communication interface, a microphone, and a processor configured to obtain a first expectation value for a first sound to be output from a first external electronic apparatus and a second expectation value for a second sound to be output from a second external electronic apparatus, control the communication interface to transmit a first sound output request signal to the first external electronic apparatus, obtain a first measurement value for the first sound, control the communication interface to transmit a second sound output request signal to the second external electronic apparatus, obtain a second measurement value for the second sound, and obtain a correction value for one or more of the first measurement value and the second measurement value based on the obtained first measurement value, the obtained second measurement value, the obtained first expectation value, and the obtained second expectation value.