摘要:
An acoustic data processor according to the present invention is used for processing acoustic data including signal sounds to reduce noises generated by a mechanical apparatus. The acoustic data processor includes a motion status obtaining section for obtaining motion status of the mechanical apparatus, an acoustic data obtaining section for obtaining acoustic data corresponding to the obtained motion status, and a database for storing various motion statuses of the mechanical apparatus in a unit time and corresponding acoustic data as templates. The acoustic data processor further includes a database searching section for searching the database to retrieve the template having the motion status closest to the obtained motion status; and a template subtraction section for subtracting the acoustic data of the template having the motion status closest to the obtained motion status from the obtained acoustic data to reduce noises generated by the mechanical apparatus.
摘要:
A sound processing device includes a storage unit configured to store first operation data corresponding to a motion of a mechanical apparatus and a first sound feature value corresponding to the motion in correlation with each other, a noise estimating unit configured to estimate a third sound feature value corresponding to a noise component based on a second sound feature value corresponding to an acquired sound signal, a sound feature value processing unit configured to calculate a target sound feature value from which the noise component is removed based on the second sound feature value and the third sound feature value, and an updating unit that updates the first sound feature value stored in the storage unit based on detected second operation data and the third sound feature value estimated by the noise estimating unit.
摘要:
Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R^) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.
摘要:
An information transmission device which analyzes a diction of a speaker and provides an utterance in accordance with the diction of the speaker, and which has a microphone detecting a sound signal of the speaker, a feature extraction unit extracting at least one feature value of the diction of the speaker based on the sound signal detected by the microphone, a voice synthesis unit synthesizing a voice signal to be uttered so that the voice signal has the same feature value as the diction of the speaker, based on the feature value extracted by the feature extraction unit, and a voice output unit performing an utterance based on the voice signal synthesized by the voice synthesis unit.
摘要:
A voice recognition system (10) for improving the toughness of voice recognition for a voice input for which a deteriorated feature amount cannot be completely identified. The system comprises at least two sound detecting means (16a, 16b) for detecting a sound signal, a sound source localizing unit (21) for determining the direction of a sound source based on the sound signal, a sound source separating unit (23) for separating a sound by the sound source from the sound signal based on the sound source direction, a mask producing unit (25) for producing a mask value according to the reliability of the separation results, a feature extracting unit (27) for extracting the feature amount of the sound signal, and a voice recognizing unit (29) for applying the mask to the feature amount to recognize a voice from the sound signal.
摘要:
A system capable of reducing the influence of sound reverberation or reflection to improve sound-source separation accuracy. An original signal X(ω,f) is separated from an observed signal Y(ω,f) according to a first model and a second model to extract an unknown signal E(ω,f). According to the first model, the original signal X(ω,f) of the current frame f is represented as a combined signal of known signals S(ω,f−m+1) (m=1 to M) that span a certain number M of current and previous frames. This enables extraction of the unknown signal E(ω,f) without changing the window length while reducing the influence of reverberation or reflection of the known signal S(ω,f) on the observed signal Y(ω,f).
摘要:
A device capable of improving the convergence rate and estimation accuracy in estimating a correlation value. According to a signal processing device, since a window length is adjusted in such a manner to reduce an estimated error of a correlation matrix, the convergence rate and estimation accuracy in estimating the correlation matrix and the correlation value as its off-diagonal element can be improved. Then, in such a high-probability condition that the correlation of plural output signals according to a state is estimated with a high degree of precision, signal processing is performed on the plural signals, so that the state can be estimated with a high degree of precision.
摘要:
An information transmission device which analyzes a diction of a speaker and provides an utterance in accordance with the diction of the speaker, and which has a microphone detecting a sound signal of the speaker, a feature extraction unit extracting at least one feature value of the diction of the speaker based on the sound signal detected by the microphone, a voice synthesis unit synthesizing a voice signal to be uttered so that the voice signal has the same feature value as the diction of the speaker, based on the feature value extracted by the feature extraction unit, and a voice output unit performing an utterance based on the voice signal synthesized by the voice synthesis unit.
摘要:
In a sound source localization system using a light emitting device for visualizing sound information, including: a light emitting device (40) including a microphone for receiving sound from a sound source (1, 2) and a light emitting means for emitting light based on the sound from the microphone; a generating section for generating light emitting information for the light emitting device (40); and a sound source localization section (60) for determining a position of the sound source based on the light emitting information from the generating section.
摘要:
A robot includes: a sound collecting unit collecting and converting a musical sound into a musical acoustic signal; a voice signal generating unit generating a self-vocalized voice signal; a sound outputting unit converting the self-vocalized voice signal into a sound and outputting the sound; a self-vocalized voice regulating unit receiving the musical acoustic signal and the self-vocalized voice signal; a filtering unit performing a filtering process; a beat interval reliability calculating unit performing a time-frequency pattern matching process and calculating a beat interval reliability; a beat interval estimating unit estimating a beat interval; a beat time reliability calculating unit calculating a beat time reliability; a beat time estimating unit estimating a beat time on the basis of the calculated beat time reliability; a beat time predicting unit predicting a beat time before the current time; and a synchronization unit synchronizing the self-vocalized voice signal.