A hearing device adapted to be worn by a user and for picking up sound containing the user's own voice is provided. The hearing device comprises a) an input unit comprising first and second input transducers for converting sound to first and second electric input signals, respectively, representing said sound; b) a processor configured to receive said first and second electric input signals and to provide a combined signal as a linear combination of the first and second electric input signals, wherein the combined signal comprises an estimate of the user's own voice, and c) wherein said hearing device is configured to provide that said first and second input transducers are located on said user at first and second locations, when worn by said user; and d) wherein said first and second locations are selected to provide that said first and second electric signals exhibit substantially different directional responses for sound from the user's mouth as well as from sound from sound sources located in an environment around the user. A method of operating a hearing device is further disclosed. Thereby an improved quality of an own voice estimate may be provided.

An apparatus comprising: an input configured to receive at least two audio signals; a frequency domain transformer configured to transform the at least two audio signals into a frequency domain representation of the at least two signals; a spatial covariance processor configured to generate an observed spatial covariance matrix from the frequency domain representations of the at least two audio signals; a beamformer configured to generate a spatial covariance matrix model comprising at least one beamformer kernel; a matrix factorizer configured to generate a linear magnitude mode! of audio objects; to combine the spatial covariance matrix model and the linear magnitude model; and further configured to determine at least one combination parameter, such that the at least one parameter for the combination attempts to optimise the combination; and a separator configured to cluster the audio objects based on the at least one combination parameter to create separated audio sources.

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.

What is described is a method for charactering an airflow, having the following steps: receiving acoustic signals generated by the airflow by means of a microphone array; extracting a characteristic information from the acoustic signals; determining an information on the airflow based on the characteristic information.

An apparatus comprising: an input configured to receive at least two groups of at least two audio signals; a first audio former configured to generate a first formed audio signal from a first of the at least two groups of at least two audio signals; a second audio former configured to generate a second formed audio signal from the second of the at least two groups of at least two audio signals; an audio analyzer configured to analyze the first formed audio signal and the second formed audio signal to determine at least one audio source and an associated audio source signal; and an audio signal synthesizer configured to generate at least one output audio signal based on the at least one audio source and the associated audio source signal.

A system can include one or more unmanned air vehicles (UAVs) that detect a gunshot sound. The one or more UAVs can confirm with each other whether or not a gunshot sound was heard and use shared information to localize the gunshot sound. Other embodiments are described.

A speech recognition method acquires sound information via multiple microphones, detects a sound source interval including sound from the sound information, acquires an estimated direction of speech by conducting direction estimation on a speech interval from among the sound source interval, conducts an adaptation process of using the sound information to estimate filter coefficients, decides a buffer size of the sound information to hold in a buffer, based on sound source interval information, estimated direction information, and adaptation process convergence state information, holds the sound information in the buffer according to the buffer size, conducts a beamforming process using the sound information held in the buffer and the filter coefficients to acquire speech information, and conducts speech recognition on the speech information acquired by the beamforming process. The method decides the buffer size to be a size sufficient for convergence of the adaptation process immediately after sound information processing starts.

A system includes an imaging part that captures an image, a sound collection part that collects sounds, a display part that displays image data captured by the imaging part, a directive direction calculation part that calculates a directive direction which directs toward a sound position corresponding to a designated position of the image data from the sound collection part when any position of the displayed image data is designated, and a control part that forms a directivity in the sounds in the calculated directive direction. The control part controls output of the sounds collected by the sound collection part or output of the sounds which are collected by the sound collection part and of which the directivity is formed, or suspends collection of sounds in the sound collection part, when it is determined that the sound position is included in a preset protection region.

An audio system that adjusts one or more beam patterns emitted by one or more loudspeaker arrays based on the preferences of users/listeners is described. The audio system includes an audio receiver that contains a listener location estimator, a listener identifier, and a voice command processor. Inputs from the listener location estimator, the listener identifier, and the voice command processor are fed into an array processor. The array processor drives the one or more loudspeaker arrays to emit beam patterns into the listening area based on inputs from each of these devices. By examining the location, preferred usage settings, and voice commands from listeners, the generated beam patterns are customized to the explicit and implicit preferences of the listeners with minimal direct input. Other embodiments are also described.

Audio pickup systems and methods are provided to enhance an audio signal by removing noise components related to an acoustic environment. The systems and methods receive a primary signal and one or more reference signals from various microphones. Adaptive filtering and combining minimizes an energy content of a resulting output signal, e.g., to form a substantially null output when the system is in a static acoustic environment. When the system is a playback sound source, one or more echo cancellers may contribute to removing content from the output signal. A change in the acoustic environment, such as a new sound source, causes content in the output signal until the adaptive filtering adapts to the new environment. In some examples, a desired content such as a wake-up word is detected and adaptation is stopped.