The inventive concept relates to a smart hearing device for providing a control parameter and feedback for a natural language or a non-natural language determined by analyzing sound data, which includes a receiving unit that receives sound data of a voice signal and a noise signal from a first microphone and a second microphone being formed at one side, a determination unit that compares digital flow of the sound data with a previously stored graph pattern to determine a natural language or a non-natural language for the sound data, a processing unit that matches similar data for the determined natural language or non-natural language, based on a database including a natural language area and a non-natural language area, and a providing unit that provides a user with a one-sided sound converted by setting a control parameter in a natural language or a non-natural language specified according to the matched similar data.

One example may include a method that includes receiving, at a server, a data set from one or more mobile devices located in a presentation space, combining the received data set with additional data to create a combined data set, creating a presentation signal based on the combined data set, subtracting a portion of one or more of the data set and the additional data set from the combined data set to create a modified presentation signal, forwarding the modified presentation signal to one or more of a display and a loudspeaker located in the presentation space, and playing the modified presentation signal via one or more of the loudspeaker and the display device.

In example implementations, an apparatus is provided. The apparatus includes a microphone, a radar, a memory, and a processor in communication with the microphone, the radar, and the memory. The microphone is to receive audio signals. The radar is to collect data on users in a location. The memory is to store known body positions associated with having a side-conversation. The processor is to determine that a user is having a side-conversation based on the data collected by the radar compared to the known body positions associated with having a side-conversation and filter noise associated with the side-conversation received by the microphone from a direction associated with the user.

An image capture device, having: a housing, a lens snout, a front microphone, a top microphone, and an audio processor. The housing has a top and front housing surface. The lens snout protrudes from the front housing surface. The front microphone mounted within or on the front housing surface and below the lens snout. The top microphone mounted within or on a top housing surface in a position biased toward the front housing surface. The audio processor comprises a memory that is configured to store instructions that when executed cause the audio processor to generate an output audio signal. The top microphone is located at a position to receive direct freestream air flow when the housing is positioned in a pitched forward orientation at a pitched forward angle relative to a vertical axis. The front microphone receives turbulent air flow from the lens snout when the housing is positioned in the pitched forward orientation.

An image capture device with beamforming for wind noise optimized microphone placements is described. The image capture device includes a front facing microphone configured to capture an audio signal. The front facing microphone co-located with at least one optical component. The image capture device further includes at least one non-front facing microphone configured to capture an audio signal. The image capture device further includes a processor configured to generate a forward facing beam using the audio signal captured by the front facing microphone and the audio signal captured by the at least one non-front facing microphone, generate an omni beam using the audio signal captured by the at least one non-front facing microphone, and output an audio signal based on the forward facing beam and the omni beam.

A wireless audio device includes a communication module, a plurality of microphones comprising an internal microphone and an external microphone, and at least one processor configured to: receive a multi-recording conversion signal from an electronic device, enable at least one microphone of the internal microphone and the external microphone based on the multi-recording conversion signal, transmit a first audio signal generated based on a sound input through the enabled at least one microphone to the electronic device, detect a microphone control input, selectively mute or unmute the at least one microphone of the internal microphone and the external microphone based on the microphone control input, and transmit, to the electronic device, a second audio signal generated based on a microphone input acquired in a state where the at least one microphone of the internal microphone and the external microphone is selectively muted or unmuted based on the microphone control input.

An apparatus including circuitry configured to: obtain at least two audio signals from at least two microphones, wherein the at least two audio signals at least in part includes noise which is substantially incoherent between the at least two audio signals; estimate values associated with the noise within the at least two audio signals; process at least one of the at least two audio signals based on the values associated with the noise; and obtain spatial metadata associated with the at least two audio signals for rendering at least one of the at least two audio signals.

A hearing device, e.g. a hearing aid, configured to be worn by a user, comprises a) two or more input transducers (e.g. microphones) wherein said two or more input transducers during use of the hearing device are arranged with a distance between them; b) a directional system comprising a directional algorithm configured to provide a directional pattern in dependence of said distance. The hearing device is configured to estimate a current value of said distance, or an equivalent acoustic delay, or beamformer weights of said directional system, thereby the directional performance can be optimized to the individual user.

Disclosed are a directional acoustic sensor, a method of adjusting directional characteristics using the directional acoustic sensor, and a method of attenuating an acoustic signal in a specific direction using the directional acoustic sensor. The directional acoustic sensor includes a plurality of resonance units arranged to have different directionalities and a signal processor configured to adjust directional characteristics by calculating at least one of a sum of and a difference between outputs of the resonance units. In this state, the signal processor attenuates an acoustic signal in a specific direction by using a plurality of directional characteristics obtained by calculating at least one of the sum of and the difference between the outputs of the resonance units at a certain ratio.

In some examples of the present disclosure, a non-transitory memory resource storing machine-readable instructions can cause a processing resource of a computing device to: instruct a first microphone of the computing device and a second microphone of a peripheral device to capture an audio signal generated by an audio source, determine a location of the second microphone based on a proximity of the peripheral device to the audio source, and alter a sound property of the audio signal based on a location of the first microphone and the location of the second microphone.