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.

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.

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.

In one embodiment, a multi-microphone system for an endpoint device receives input signals for a remote conference between the endpoint device and at least one other endpoint device. The multi-microphone system may include at least a top microphone unit and a bottom microphone unit. A signal degradation event that causes degradation of signals received by the top microphone unit or the bottom microphone unit is detected. Then, based on information regarding the signal degradation event, it is determined whether the signal degradation event affects one or both of the top microphone unit and the bottom microphone unit. In response, an output signal is generated for transmission to the at least one other endpoint device, and the output signal uses a portion of the input signals that excludes signals received by the top microphone unit and/or the bottom microphone unit determined to be affected by the signal degradation event.

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.

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.

Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by two or more network microphone devices include receiving an instruction to process one or more audio signals captured by a first network microphone device and after receiving the instruction (i) disabling at least a first microphone of a plurality of microphones of a second network microphone device, (ii) capturing a first audio signal via a second microphone of the plurality of microphones, (iii) receiving over a network interface of the second network microphone device a second audio signal captured via at least a third microphone of the first network microphone device, (iv) using estimated noise content to suppress first and second noise content in the first and second audio signals, (v) combining the suppressed first and second audio signals into a third audio signal, and (vi) determining that the third audio signal includes a voice input comprising a wake word.

A remote microphone device for a hearing aid system comprises a multitude of microphones providing a corresponding multitude of electric input signals; a digital signal processor for providing a processed signal in dependence of said multitude of electric input signals. The processor comprises a noise reduction system comprising at least one beamformer for providing a spatially filtered signal. The remote microphone device further comprises a wireless communication interface comprising antenna and transceiver circuitry allowing the remote microphone device to transmit said processed signal comprising said spatially filtered signal or a further processed version thereof to said hearing aid system; a rechargeable battery; a casing having a planar structure with a form and size as a standard credit card; and a user interface configured to allow the user to control functionality of the remote microphone device. The invention may e.g. be used in hearing aid systems or headsets.

An example conference speech enhancement method includes obtaining information about a sound pickup area and a location relationship between a first microphone array and a second microphone array. A relative location relationship can then be obtained between a sound source and each of the first microphone array and the second microphone array. Location information of the sound source can then be determined based on the location relationship and the relative location relationship. In response to determining that the sound source is located in the sound pickup area, enhancing a sound signal corresponding to the sound source.

Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by two or more network microphone devices include receiving an instruction to process one or more audio signals captured by a first network microphone device and after receiving the instruction (i) disabling at least a first microphone of a plurality of microphones of a second network microphone device, (ii) capturing a first audio signal via a second microphone of the plurality of microphones, (iii) receiving over a network interface of the second network microphone device a second audio signal captured via at least a third microphone of the first network microphone device, (iv) using estimated noise content to suppress first and second noise content in the first and second audio signals, (v) combining the suppressed first and second audio signals into a third audio signal, and (vi) determining that the third audio signal includes a voice input comprising a wake word.