摘要:
Disclosed are an apparatus and a method for beamforming in consideration of characteristics of an actual noise environment. The apparatus includes a microphone array having at least microphone, the microphone array outputting a signal input through the microphone; a coherence function generation unit for calculating coherences for input signals according to each space between microphones, calculating averages of the coherences for the same distance, and filtering the calculated averages of the coherences and outputting the resultant values, when an input signal is input; a spatial filter factor calculation unit for calculating and outputting a spatial filter factor by using the filtered average coherences; and a beamforming execution unit for performing a beamforming for the input signals by using the spatial filter factor, thereby outputting a noise-processed signal.
摘要:
Disclosed are an apparatus and a method for beamforming in consideration of characteristics of an actual noise environment. The apparatus includes a microphone array having at least microphone, the microphone array outputting a signal input through the microphone; a coherence function generation unit for calculating coherences for input signals according to each space between microphones, calculating averages of the coherences for the same distance, and filtering the calculated averages of the coherences and outputting the resultant values, when an input signal is input; a spatial filter factor calculation unit for calculating and outputting a spatial filter factor by using the filtered average coherences; and a beamforming execution unit for performing a beamforming for the input signals by using the spatial filter factor, thereby outputting a noise-processed signal.
摘要:
Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.
摘要:
Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.
摘要:
A system and method for sound source separation. The system and method use a beamforming technique. The sound source separation system includes a windowing processor; a DFT transformer; a transfer function estimator; and a noise estimator. The system also includes a voice signal extractor that cancels individual voice signals, except an individual voice signal that is desired to be extracted among individual voice signals, from the integrated voice signals. The system further includes a voice signal detector that cancels a noise part provided through the noise estimator from a transfer function of an individual voice signal which is desired to be detected and extracts a noise-canceled individual voice signal. Even when two or more sound sources are simultaneously input, the sound sources can be separated from each other and separately stored and managed, or an initial sound source can be stored and managed.
摘要:
An adaptive mode control apparatus and method for adaptive beamforming based on detection of a user direction sound are provided. The adaptive mode control apparatus includes a signal intensity detector that searches for signal intensity of each designated direction to detect signal intensity having a maximum value when a voice signal of each direction is input through at least one microphone; and an adaptive mode controller that compares the signal intensity having the maximum value detected through the signal intensity detector with a threshold value and determines whether to perform an adaptive mode of a Generalized Sidelobe Canceller (GSC) according to the comparison results. Therefore, a lack of control of adaptation of an adaptive filter of the conventional art is solved. That is, as one condition for guaranteeing performance of adaptive beamforming, adaptation of an adaptive filter is not performed when noise of a sound with a high autocorrelation is cancelled.
摘要:
A system and method for sound source separation. The system and method use a beamforming technique. The sound source separation system includes a windowing processor; a DFT transformer; a transfer function estimator; and a noise estimator. The system also includes a voice signal extractor that cancels individual voice signals, except an individual voice signal that is desired to be extracted among individual voice signals, from the integrated voice signals. The system further includes a voice signal detector that cancels a noise part provided through the noise estimator from a transfer function of an individual voice signal which is desired to be detected and extracts a noise-canceled individual voice signal. Even when two or more sound sources are simultaneously input, the sound sources can be separated from each other and separately stored and managed, or an initial sound source can be stored and managed.