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.

An apparatus is configured to obtain a plurality of phase differences, wherein each phase difference represents a difference in phase between a spectral component in a first sound signal and a corresponding spectral component in a second sound signal, to estimate an aliasing frequency, wherein phase differences in frequency bands above the aliasing frequency are expected to be wrapped, and to unwrap the plurality of phase differences in dependence on the estimated aliasing frequency, wherein a phase difference obtained from spectral components that are comprised in one frequency band is unwrapped independently of any phase difference obtained from spectral components that are comprised in another frequency band.

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 method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction.

Provided is a source localization in an apparatus for performing a source localization, a target sound source enhancement or speech recognition. The source localization method using input signals input from a plurality of microphones, comprising steps of: (a) obtaining a log likelihood function or an auxiliary function under the assumption that a target source signal mixed with noises satisfies a CGMM model; (b) obtaining an equation for estimating parameter values of the log likelihood function or the auxiliary function so that a value of the log likelihood function or the auxiliary function is maximized recursively in each time frame; (c) estimating a covariance matrix recursively in each time frame; and (d) estimating a steering vector recursively by using the estimated covariance matrix, wherein the steering vector of the target sound source is estimated from the input signals.

Provided is a sound source localization method including steps of: (a) receiving a mixed signal of a target sound source signal and noise and echo signals through multiple microphones including at least two microphones; (b) generating a binarized mask based on a diffuseness by using a coherence-to-diffuseness ratio CDR, which is information on the target sound source and the noise source, by using the input signal; (c) pre-processing an input signal to multiple microphones by using the generated binarized mask; and (d) performing a predetermined algorithm such as the GCC-PHAT or the SRP-PHAT on the pre-processed input signal to estimate a direction of the target sound source.

A method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction.

A hearing device comprises a sound system for estimating the direction of arrival of sound emitted by one or more sound sources creating a sound field. The sound system comprises an array of N sound receiving transducers (microphones), each providing an electric input signal, a processing unit comprising a) a model unit comprising a parametric model configured to be able to describe the sound field at the array as a function of the direction of arrival in a region surrounding and adjacent to the array; b) a model optimizing unit configured to optimize said model with respect to its parameters based on said sound samples; c) a cost optimizing unit configured to minimize a cost function of the model with respect to said direction of arrivals; d) an estimating unit configured to estimate the direction of arrival based on said parametric model with the optimized parameters and the optimized cost function.

A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialogue phase.

A speech recognition device is provided. The speech recognition device includes at least one microphone configured to receive a sound signal from a first sound source, and at least one processor configured to determine a direction of the first sound source based on the sound signal, determine whether the direction of the first sound source is in a registered direction, and based on whether the direction of the first sound source is in the registered direction, recognize a speech from the sound signal regardless of whether the sound signal comprises a wake-up keyword.