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 minimise the power of the output signal.

Methods and systems are provided for receiving desired sounds. The system includes a position sensor configured to determine an occupant position of an occupant engaging in speech within a defined space and transmit the speaking occupant position. A plurality of microphones are configured to receive sound from within the defined space and transmit audio signals corresponding to the received sound. A processor, in communication with the position sensor and the microphones, is configured to receive the speaking occupant position and the audio signals, apply a beamformer to the audio signals to direct a microphone beam toward the occupant position, and generate a beamformer output signal.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

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.

A microphone array is a microphone array including two or more microphone holders 1. The two or more microphone holders 1 each hold one microphone. The two or more microphone holders 1 are separatable from each other. The two or more microphone holders 1 are combinable in two or more combination manners different from each other such that the two or more microphone holders 1 are each in contact with any one of the other microphone holders 1 and a distance between the two respective microphones held by two of the microphone holders 1 that are in contact with each other is a predetermined distance.

To perform an efficient and stable area sound pick-up process. The present invention relates to a sound pick-up apparatus. The sound pick-up apparatus according to the present invention includes: a means for acquiring target direction signals on the basis of beamformers of input signals supplied by a plurality of microphone arrays; a means for calculating correction coefficients for approximating target area sound components to each other, the target area sound components being included in the respective target direction signals of the plurality of microphone arrays; a means for selecting a main microphone array on the basis of the correction coefficients, the main microphone array being to be used as a criterion for extracting target area sound; and a means for correcting the target direction signals of the respective microphone arrays by using the correction coefficients with respect to the main microphone array, and extracting the target area sound on the basis of the corrected target direction signals of the respective microphone arrays.

A system for directionally selective sound reception comprises an array of pressure sensors (120a, 120c) each arranged to output a pressure signal indicative of pressure, and a processor arranged to receive the pressure signals. The sensor array comprises a support (130) supporting the four sensors. Two of the sensors are mounted on one side of the support and at least a third sensor is supported on an opposite side of the support. The sound pressure difference measured between the first sensor and the second sensor caused by sound arriving at the array from a direction parallel to the support (130) is dependent on the distance between the first and second sensors and the nature of material in the space between the first and second sensors. The sound pressure difference measured between the first and third sensors caused by sound travelling perpendicular to the support is dependent on the distance between the first and third sensors. The nature of material in the space between the first and third sensors, and the spacings and the materials are selected such that the sound pressure differences are substantially equal.

There are provided an ultrasound image generating apparatus, which generates a high-quality ultrasound image even in a deep portion of a subject, and a control method thereof. In an ultrasound image (Img), for a portion (Ar1) equal to or less than a depth threshold value (D1), a real scanning line (L1) obtained from an acoustic wave echo signal is used. In the ultrasound image (Img), for a portion (Ar2) deeper than the depth threshold value (D1), an interpolation scanning line (L2) located between the real scanning lines (L1) is generated from an acoustic wave echo signal having a positional deviation between a focusing position of ultrasound waves and an observation target position. Also for a portion deeper than the interpolation scanning line (L2), a high-quality ultrasound image (Img) is obtained.

Methods and system are described for cancelling interference in a microphone system. A positive bias voltage is applied to a first microphone diaphragm and a negative bias voltage is applied to a second microphone diaphragm. The diaphragms are configured to exhibit substantially the same mechanical deflection in response to acoustic pressures received by the microphone system. A differential output signal is produced by combining a positively-biased output signal from the first microphone diaphragm and a negatively-biased output signal from the second microphone diaphragm. This combining cancels common-mode interferences that are exhibited in both the positively-biased output signal and the negatively-biased output signal.

A personal audio device configured to be worn on the head or body of a user and including a plurality of microphones configured to provide a plurality of separate microphone signals capturing audio from an environment external to the personal audio device, and a processor configured to process a first subset of the plurality of separate microphone signals using a first array processing technique to provide a first array signal, compare the first array signal to a microphone signal from the plurality of separate microphone signals, and select the first array signal or the microphone signal based on the comparison.