A sound processing apparatus includes a sound collection position calculating unit configured to calculate sound collection positions of sound signals of multiple channels on the basis of the sound signals, a sound source direction calculating unit configured to calculate a sound source direction on the basis of the sound signals of multiple channels, a first transfer function calculating unit configured to calculate a first transfer function corresponding to the sound source direction on the basis of the sound signals of multiple channels, and a second transfer function calculating unit configured to calculate a second transfer function by interpolating the first transfer function corresponding to each of a plurality of sound source directions.

[Object] To enable both a reduction in the number of sound collection portions and an improvement in a resolution in estimation of a direction of a sound source to be compatible.

[Solution] An information processing device including: an acquisition unit configured to acquire a sound collection result of a sound from each of one or more sound sources obtained by a sound collection portion of which positional information indicating at least one of a position and a direction is changed; and an estimation unit configured to estimate a direction of each of the one or more sound sources on a basis of a change in a frequency of a sound collected by the sound collection portion in association with a change in the positional information of the sound collection portion.

In accordance with embodiments of the present disclosure, a method for detecting near-field sources in an audio device may include computing a normalized cross correlation function between a first microphone signal and a second microphone signal, computing normalized auto correlation functions of each of the first microphone signal and the second microphone signal, partitioning the normalized cross correlation function and the normalized auto correlation functions into a plurality of time lag regions, computing for each respective time lag region of the plurality of the time lag regions a respective maximum deviation between the normalized cross correlation function and a normalized auto correlation function within the respective time lag region, combining the respective maximum deviations from the plurality of time lag regions to derive multiple detection statistics, and comparing each detection statistic of the multiple detection statistics to a respective threshold to detect a near-field signal.

A sound source identification apparatus includes a sound collection unit including a plurality of microphones, a sound source localization unit configured to localize a sound source on the basis of an acoustic signal collected by the sound collection unit, a sound source separation unit configured to perform separation of the sound source on the basis of the signal localized by the sound source localization unit, and a sound source identification unit configured to perform identification of a type of sound source on the basis of a result of the separation in the sound source separation unit, and a signal input to the sound source identification unit is a signal having a magnitude equal to or greater than a first threshold value which is a predetermined value.

A method for displaying a user interface on an electronic device is described. The method includes presenting a user interface. The user interface includes a coordinate system. The coordinate system corresponds to physical coordinates based on sensor data. The method also includes providing a sector selection feature that allows selection of at least one sector of the coordinate system. The method further includes providing a sector editing feature that allows editing the at least one sector.

The proposed method for localizing a target sound source from a plurality of sound sources, wherein a multi-channel recording signal of the plurality of sound sources comprises a plurality of microphone channel signals, comprises converting each microphone channel signal into a respective channel spectrogram in a time-frequency domain, blindly separating the channel spectrograms to obtain a plurality of separated source signals, identifying, among the plurality of separated source signals, the separated source signal that best matches a target source model, estimating, based on the identified separated source signal, a binary mask reflecting where the target sound source is active in the channel spectrograms in terms of time and frequency, applying the binary mask on the channel spectrograms to obtain masked channel spectrograms, and localizing the target sound source from the plurality of sound sources based on the masked channel spectrograms.

A method of processing an acoustic image includes the steps of acquiring acoustic signals generated by acoustic sources in a predetermined region of space, generating a multispectral 3D acoustic image that includes a collection of 2D acoustic images, performing a frequency integration of the multispectral acoustic image for generating a 2D acoustic map, locating at least one target acoustic source of interest and modeling the signal spectrum associated with the target acoustic source, generating a classification map obtained by comparing the signal spectrum of each signal associated with each pixel of the multispectral acoustic image and the model of the signal spectrum associated with the target acoustic source to distinguish the spectrum of the signal associated with the target acoustic source from the signal spectra associated with the remaining acoustic sources, and merging the classification map and the acoustic map to obtain a merged map.

The present invention includes methods, circuits, devices, systems and associated computer executable code for acquiring, processing and rendering acoustic signals. According to some embodiments, one or more direction specific audio signals may be generated using a microphone array comprising two or more microphones and an audio stream generator. The audio stream generator may receive a direction parameter from an optical tracking system. There may be provided an audio rendering system adapted to normalize and/or balance acoustic signals acquired from a soundscape.

A voice processing device includes: a sound source localization unit configured to determine a direction of each sound source on the basis of voice signals of a plurality of channels; a sound source separation unit configured to separate signals for respective sound sources indicating components of respective sound sources from the voice signals of the plurality of channels; a speech section detection unit configured to detect a speech section in which the number of speakers is 1 from the signals for respective sound sources; and a speaker identification unit configured to identify a speaker on the basis of the signals for respective sound sources in the speech section.

An audio processing device includes a sound source localization unit that determines respective directions of sound sources from audio signals of a plurality of channels, a setting information selection unit that selects a setting information from a setting information storage unit that stores setting information including transfer functions of directions in advance for each acoustic environment, and a sound source separation unit that separates the audio signals of the plurality of channels into respective sound-source-specific signals of sound sources by applying a separation matrix based on transfer functions included in the setting information selected by the setting information selection unit.