A mobile platform for calibrated data acquisition includes a transceiver within the mobile platform, a locomotion unit configured to move the mobile platform within an area, a sensor coupled with the locomotion unit and configured to output a signal, and a controller that is configured to request a measurement of a parameter from the sensor, remove from the measurement, background noise associated with the mobile platform, thereby focusing the measurement to foreground noise, in response to the mobile platform reaching a new position and direction, request a second measurement of the parameter from the sensor, remove from the second measurement, background noise associated with the mobile platform at the new position, aggregate the signal from the sensor and associated position and direction to create an energy map via spatio-dynamic beamforming, and analyzing the energy map to identify a state of an apparatus in the area.

This disclosure describes an apparatus and method of an embodiment of an invention that is a band-limited beamforming microphone array with acoustic echo cancellation that includes: a plurality of first microphones configured as a beamforming microphone array to resolve first audio input signals within a first frequency range, the beamforming microphone array includes acoustic echo cancellation; one or more additional microphone(s) configured to resolve second audio input signals within a restricted second frequency range such that the additional microphone(s) are coupled to the beamforming microphone array; augmented beamforming that processes audio signals from the beamforming microphone array and the additional microphone(s).

Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.

An apparatus comprising means for: capturing an audio scene using multiple microphones; determining that the captured audio scene has unacceptable detected noise; in dependence upon determining that the captured audio scene has unacceptable detected noise, searching both different sets of one or more microphones and different physical rotation angles of the microphones to find a combination of a first set of one or more microphones and a first physical rotation angle of the microphones that captures the audio scene with acceptable detected noise; and controlling the physical rotation angle of the microphones to be the first physical rotation angle of the microphones and capturing the audio scene using the combination of the first set of one or more microphones and the first physical rotation angle of the microphones.

An apparatus and method for real-time audio processing employs a gaze detection sensor to detect a direction of a user's gaze and output a gaze signal corresponding to the detected direction of the user's gaze. A digital signal processing unit responds to a plurality of signals corresponding to a plurality of sounds received at the apparatus, and the determined direction of gaze to identify a signal of interest from the plurality of signals using the gaze signal. The signal of interest is processed for output to the user. In embodiments, a microphone array provides the plurality of signals. An imaging sensor may work with either the microphone array or the gaze detection sensor to identify the signal of interest.

The application relates to an audio processing system and a method of processing a noisy (e.g. reverberant) signal comprising first (v) and optionally second (w) noise signal components and a target signal component (x), the method comprising a) Providing or receiving a time-frequency representation Y.sub.i(k,m) of a noisy audio signal y.sub.i at an i.sup.th input unit, i=1, 2, . . . , M, where M≧2; b) Providing (e.g. predefined spatial) characteristics of said target signal component and said noise signal component(s); and c) Estimating spectral variances or scaled versions thereof λ.sub.V, λ.sub.X of said first noise signal component v (representing reverberation) and said target signal component x, respectively, said estimates of λ.sub.V and λ.sub.X being jointly optimal in maximum likelihood sense, based on the statistical assumptions that a) the time-frequency representations Y.sub.i(k,m), X.sub.i(k,m), and V.sub.i(k,m) (and W.sub.i(k,m)) of respective signals y.sub.i(n), and signal components x.sub.i, and v.sub.i (and w.sub.i) are zero-mean, complex-valued Gaussian distributed, b) that each of them are statistically independent across time m and frequency k, and c) that X.sub.i(k,m) and V.sub.i(k,m) (and W.sub.i(k,m)) are uncorrelated. An advantage of the invention is that it provides the basis for an improved intelligibility of an input speech signal. The invention may e.g. be used for hearing assistance devices, e.g. hearing aids.

Problem to be Solved To effectively operate a microphone array provided with a sound source position estimating function. Solution A microphone control system of the present invention is realized by the use of a tablet-type computer, for example. On the display of the control system, a two-dimensional simulation diagram 302, for example, simulating the space in which the microphone array is disposed. An array mark 304 simulating the exterior shape of the microphone array is displayed on the simulation diagram 302. Further, a sound source mark 316 indicating the position of the sound source estimated by the microphone array is displayed on the simulation diagram 302. Thus, an operator can intuitively find the position of the sound source in the space in which the microphone array is disposed, by referring to the simulation diagram 302, the array mark 304 and the sound source mark 316 displayed on the simulation diagram. Thus, the usage of the microphone array is improved.

Embodiments include a method comprising forming a pipe including a first end and a second end by forming couplings between a plurality of sections of pipe. A loudspeaker is connected to the first end. The loudspeaker is a mouth simulator loudspeaker. A plurality of microphones are positioned a third distance inside an inside surface of the pipe using a receptacle positioned in the pipe a first distance from the first end and a second distance from the second end. An acoustic output is generated at the loudspeaker. One or more calibration filters are generated using outputs of the plurality of microphones produced in response to the acoustic output.

An apparatus comprising: a detector configured to determine at least one microphone is impaired by analysing at least one audio signal from the at least one microphone; and an controller configured to determine an indicator based on the determination of the impairment of the at least one microphone; and configured to apply the indicator based on the determination of the impairment of the at least one microphone, such that the at least one audio signal is processed based on the indicator.

A method and apparatus are provided for processing data for estimating mixing parameters of at least one audio spot signal captured by a sound recording device, called a spot microphone, arranged in the vicinity of a source among a plurality of acoustic sources constituting a sound scene, and a primary audio signal captured by an ambisonic sound recording device, arranged to capture said plurality of acoustic sources of the sound scene.