A method for determining microphone position is a method for determining positions of a plurality of microphones in a microphone array having the plurality of microphones arranged in a plurality of concentric circles. The method for determining microphone position includes a constraint condition acquiring step of acquiring constraint conditions including the maximum number of the plurality of microphones; and a selecting step of selecting, from among a plurality of combinations of (i) the number of microphones included in each of the plurality of concentric circles and (ii) the radius of each of the plurality of concentric circles, a combination indicating directional characteristics with the smallest difference from a target value of the directional characteristics of the microphone array, where the plurality of combinations satisfy the constraint conditions.

The interactive aircraft cabin environment control system employs at least one microphone array disposed within the cabin to capture spoken utterances from a passenger and is configured to provide an estimation of passenger location within the cabin based on arrival time analysis of the spoken utterances. A data source onboard the aircraft provides flight context information. Such data sources include sensors measuring real-time parameters on the aircraft, the current flight plan of the aircraft, singly and in combination. A control processor, coupled to the microphone array, is configured to ascertain passenger identity based on the spoken utterances. The control processor is programmed and configured to learn and associate passenger preference to passenger identity. The control processor is receptive of the estimation of passenger location and is coupled to provide supervisory control over at least one device forming a part of the cabin environment according to passenger location, passenger preference obtained from passenger identity and flight context information.

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.

Method and apparatus for automatic gain control utilizing sound source position information in a shared space having a plurality of microphones and a plurality of sound sources. Sound signals are received from the microphones. One or more processors locate position information corresponding to each of the sound sources. The processor(s) determine the distance to each of the sound sources from each of the microphones. The processor(s) define a predetermined gain weight adjustment for each of the microphones. The processor(s) apply the defined weight adjustments to the microphones to achieve a consistent volume of the desired plurality of sound sources. The processor(s) maintain a consistent ambient sound level regardless of the position of the sound sources and the applied gain weight adjustments. The processor(s) output a summed signal of the sound sources at a consistent volume with a constant ambient sound level across the plurality of sound source positions.

An example method of operation may include initializing a microphone array in a defined space to receive one or more sound instances based on a preliminary beamform tracking configuration, detecting the one or more sound instances within the defined space via the microphone array, modifying the preliminary beamform tracking configuration, based on a location of the one or more sound instances, to create a modified beamform tracking configuration, and saving the modified beamform tracking configuration in a memory of a microphone array controller.

This disclosure describes a conferencing device with beamforming and echo cancellation that includes: a microphone array that further comprises a plurality of microphones oriented to develop a corresponding plurality of microphone signals; a processor configured to execute the following steps: (1) performing a beamforming operation to combine the plurality of microphone signals from the microphone array into a plurality of combined signals, (2) performing an acoustic echo cancellation operation on the plurality of combined signals to generate a plurality of combined echo cancelled signals, (3) receiving with a voice activity detector the far end signal as an input, (4) selecting one or more of the combined echo cancelled signals for transmission to the far end where a signal selector uses the far end signal as information to inhibit the signal selector from changing the selection of the combined echo cancelled signals while only the far end signal is active.

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 transducer arrangement for head- or earphones comprises a sound steering unit, which comprises a frontal chamber and at least one sound canal. Each of the at least one sound canal comprises at least one internal opening towards the frontal chamber and an external open end for directing sound towards the outside of the transducer arrangement. The transducer arrangement further comprises a rear chamber and at least one loudspeaker arranged between the frontal chamber and the rear chamber.

The aircraft intercom employs beam steerable microphone arrays and speaker arrays deployed in each of a plurality of zones within the aircraft. A speech recognizer generates recognition text from utterances picked up by the microphone arrays. A direction control processor analyzes the arrival time of signals from the microphone arrays to identify utterance location, which is then used to control the beam direction of at least one steerable speaker array. A dialogue manager processor coupled to the speech recognizer and to each of the plurality of microphone arrays and plurality of speaker arrays responds to a set of predefined keywords to selectively route a voiced communication from a first selected zone to a second selected zone using the microphone array and speaker array disposed in each of the first and second selected zones.

An ear-mountable listening device includes an adaptive phased array of microphones, a speaker, and electronics. The microphones are physically arranged into a ring pattern to capture sounds emanating from an environment. Each of the microphones is configured to output one of a plurality of first audio signals that is representative of the sounds captured by a respective one of the microphones. The speaker is arranged to emit audio into an ear. The electronics are coupled to the adaptive phased array and the speaker and include logic that when executed causes the ear-mountable listening device receive a user input identifying a first sound for cancelling or amplifying, steer a null or a lobe of the adaptive phased array based upon the user input, and generate a second audio signal that drives the speaker based upon a combination of one or more of the first audio signals.