摘要:
An apparatus and method for calibrating gain difference between microphones included in a microphone array are provided. In the gain calibrating apparatus, weights for each frequency component of the acoustic signals, which have been converted into the signals in the frequency domain are calculated. The weights are used to calibrate the acoustic signals such that the plurality of acoustic signals each have the same amplitude while the acoustic signals maintain their individual phase. The amplitudes of the acoustic signals are calibrated by use of the calculated weights. The gain calibrating apparatus calibrates gain in real time while calculating weights for frequency components of the frame of acoustic signals in real time.
摘要:
A formant frequency estimation method which is important information in speech recognition by accelerating a spectrum using a pitch frequency, and an apparatus using the method is provided. That is, the formant frequency estimation method includes preprocessing an input speech signal and generating a spectrum by a fast Fourier transforming the preprocessed input speech signal; smoothing the generated spectrum; accelerating the smoothed spectrum; and determining a formant frequency on the basis of the accelerated spectrum.
摘要:
A formant frequency estimation method which is important information in speech recognition by accelerating a spectrum using a pitch frequency, and an apparatus using the method is provided. That is, the formant frequency estimation method includes preprocessing an input speech signal and generating a spectrum by a fast Fourier transforming the preprocessed input speech signal; smoothing the generated spectrum; accelerating the smoothed spectrum; and determining a formant frequency on the basis of the accelerated spectrum.
摘要:
A microphone array apparatus is provided. The microphone array apparatus includes a plurality of first type microphones which are disposed to be hidden in a direction of a target source and surrounded by a cover that passes acoustic signals therethrough, and a plurality of second type microphones which are disposed at both sides of the plurality of first type microphones.
摘要:
Provided are an apparatus and method for estimating noise that changes with time. The apparatus may calculate a speech absence probability that indicates the possibility of the absence of speech in each frequency component of an input acoustic signal, may discriminate between a speech-dominant region and a noise region from the acoustic signals based on the speech absence probability, and may estimate noise according to the discrimination result.
摘要:
An apparatus and method for calibrating gain difference between microphones included in a microphone array are provided. In the gain calibrating apparatus, weights for each frequency component of the acoustic signals, which have been converted into the signals in the frequency domain are calculated. The weights are used to calibrate the acoustic signals such that the plurality of acoustic signals each have the same amplitude while the acoustic signals maintain their individual phase. The amplitudes of the acoustic signals are calibrated by use of the calculated weights. The gain calibrating apparatus calibrates gain in real time while calculating weights for frequency components of the frame of acoustic signals in real time.
摘要:
An audio quality enhancing apparatus and method is provided in which a microphone array has a non-uniform configuration and thus a beam pattern of a desired direction is obtained in a wide range of frequencies including higher frequency bands and lower frequency bands even when the microphone array is relatively small. The audio quality enhancing apparatus includes at least three microphones which are disposed in a non-uniform configuration, a frequency conversion unit configured to transform acoustic signals input from the at least three microphones to acoustic signals of frequency domain; a band division and merging unit configured to divide frequencies of the transformed acoustic signals into bands based on intervals between the at least three microphones and to merge the acoustic signals in the frequency domain into signals of two channels based on the divided frequency bands; and a two channel beamforming unit configured to reduce noise of signals including input from a direction other than the direction of a target sound by performing beamforming on the signals of the two channels and to output the noise-reduced signals.
摘要:
A method and apparatus of estimating a voicing for speech recognition by using local spectral information. The voicing estimation method for speech recognition includes performing a Fourier transform on input voice signals after performing pre-processing on the input voice signals. The method further includes detecting peaks in the input voice signals after smoothing the input voice signals. The method also includes computing every frequency bound associated with the detected peaks, and determining a class of a voicing according to each computed frequency bound.
摘要:
A method of measuring confidence of speech recognition in a speech recognizer compares a phase change point with a phoneme string change point and uses a difference between the phase change point and the phoneme string change point and a likelihood ratio, and an apparatus using the method is provided. That is, the method of the present invention includes detecting a phase change point of a speech signal; detecting a phoneme string change point according to a result of speech recognition; calculating confidence of the speech recognition by using a difference between the detected phase change point and phoneme string change point. According to the present invention, a performance of measuring confidence may become improved by simultaneously taking not only a likelihood ratio, but also taking a comparison result of a phase change point with a phoneme string change point into consideration.
摘要:
A method and apparatus of estimating a voicing for speech recognition by using local spectral information. The voicing estimation method for speech recognition includes performing a Fourier transform on input voice signals after performing pre-processing on the input voice signals. The method further includes detecting peaks in the input voice signals after smoothing the input voice signals. The method also includes computing every frequency bound associated with the detected peaks, and determining a class of a voicing according to each computed frequency bound.