A headset implements an earloop microphone and includes a housing. An earloop of the headset secures the headset to an ear of a user. A first microphone is acoustically coupled to a first opening in the housing. A second microphone is acoustically coupled to a second opening in the housing. A third microphone is acoustically coupled to a third opening in the earloop.

Array microphone systems and methods having adjustable lobe shapes are provided. The lobe shapes of pickup patterns in an array microphone may be adjusted by weighting the audio signals of subsets of the microphone elements that make up the array. The lobe shapes may be adjusted in a direction independent of a steering vector of the lobe. Users may have greater control of lobes which can result in more efficient and optimal coverage of audio sources in environments.

Systems and methods for improving performance of a directional audio capture system are provided. An example method includes correlating phase plots of at least two audio inputs, with the audio inputs being captured by at least two microphones. The method can further include generating, based on the correlation, estimates of salience at different directional angles to localize a direction of a source of sound. The method can allow providing cues to the directional audio capture system based on the estimates. The cues include attenuation levels. A rate of change of the levels of attenuation is controlled by attack and release time constants to avoid sound artifacts. The method also includes determining a mode based on an absence or presence of one or more peaks in the estimates of salience. The method also provides for configuring the directional audio capture system based on the determined mode.

An acoustic device that is adapted to be worn on the body of a user, with a first acoustic transducer and a second acoustic transducer, where the first transducer is closer to the expected location of a first ear of the user than is the second transducer, a third acoustic transducer and a fourth acoustic transducer, where the third transducer is closer to the expected location of a second ear of the user than is the fourth transducer, and a controller that is adapted to independently control the phase and frequency response of the first, second, third and fourth transducers.

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.

Wind noise reduction in microphone signals. A first microphone signal is obtained from a first omnidirectional microphone and, contemporaneously, a second microphone signal is obtained from a second omnidirectional microphone. The first and second microphone signals are mixed to produce an output signal. Mixing involves weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals, and summing the first and second weighted microphone signals together to produce the output signal. The first and second signal weights are calculated to minimize the power of the output signal.

A system includes a headset to capture sound and a visual signal of a local area including one or more sound sources. The system determines a strength of the audio signal and a portion of the visual signal associated with the audio signal, compares the strengths, selects the weaker signal, and augments the weaker signal. The headset accordingly presents augmented audio-visual content to a user, thereby enhancing the user's perception of the weak signal.

Embodiments include an audio system comprising a speaker array comprising a plurality of drivers arranged in a first planar configuration; a microphone array comprising a plurality of transducers arranged in a second planar configuration; and a beamformer communicatively coupled to the speaker array and the microphone array, the beamformer configured to: generate an individual speaker output signal for each of the drivers in the speaker array based on one or more input audio signals received from an audio source, and generate a microphone output signal for the microphone array based on one or more microphone signals captured by one or more of the transducers. Embodiments also include a method performed by one or more processors coupled to a microphone array having a plurality of transducers and a speaker array having a plurality of drivers.

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments.