• 음성과학
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• 제4권2호
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• pp.41-54
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• 1998

Speech signals are, depending on the characteristics of waveform, classified as voiced sound, unvoiced sound, and silence. Voiced sound, produced by an air flow generated by the vibration of the vocal cords, is quasi-periodic, while unvoiced sound, produced by a turbulent air flow passed through some constriction in the vocal tract, is noise-like. Silence represents the ambient noise signal during the absence of speech. The need for deciding whether a given segment of a speech waveform should be classified as voiced, unvoiced, or silence has arisen in many speech analysis systems. In this paper, a voiced/unvoiced/silence classification algorithm using spectral change in the wavelet transformed signal is proposed and then, experimental results are demonstrated with our discussions.

• 한국음향학회지
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• 제4권1호
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• pp.9-15
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• 1985

In this paper, we describe a new algorithm for deciding whether a given segment of a speech signal is classified as voiced speech, unvoiced speech, or silence, based on parameters made on the signal. The measured parameters for the voiced-unvoiced classfication are the areas of each Zero crossing interval, which is given by multiplication of the magnitude by the inverse zero corssing rate of speech signals. The employed parameter for the unvoiced-silence classification, also, are each of positive area summation during four milisecond interval for the high frequency emphasized speech signals.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1984년도 추계학술발표회 논문집
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• pp.73-77
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• 1984

This paper reports on a Voiced-Unvoiced-Silence Classification of speech for Korean Speech Recognition. In this paper, it is describe a method which uses a Pattern Recognition Technique for classifying a given speech segment into the three classes. Best result is obtained with the combination using ZCR, P1, Ep and classification error rate is less than 1%.

• 대한전자공학회논문지SP
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• 제45권5호
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• pp.111-117
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• 2008

본 논문에서는 패턴 인식에서 우수한 성능을 보이는 가우시안 혼합모델 (Gaussian mixture model, GMM)을 이용하여 비정상적인 잡음환경에서 3GPP2 selectable mode vocoder (SMV)의 유/무성음 분류 알고리즘 성능 향상을 위한 방법을 제안한다. 기존의 SMV에 대해서 분석하고, 이론 기반으로 유/무성음 분류 알고리즘에서 우수한 성능을 보여주는 특징 벡터를 선택하여 GMM의 입력벡터로 효과적으로 이용한다 다양한 잡음환경에서 시스템의 성능을 평가한 결과 GMM을 이용한 제안된 방법이 기존의 SMV의 방법보다 우수한 유/무성음 분류 성능을 보였다.

• ETRI Journal
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• 제43권1호
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• pp.82-94
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• 2021

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
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• pp.1615-1618
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• 1987

시간 영역에서 만들어진 음성신호의 파라미터을 이용하여 주어진 음성신호의 구간이 유성음, 무성음, 혹은 묵음인지를 분류하는 새로운 알고리듬을 제시하였다. 이에 사용한 파라미터은 구간내에서 샘플링된 값의 절대치 합과 일정한 level 이상의 peak의 합(T-peak), T-peak와 절대치 합의 비 그리고, DPCM의 절대치 합들이다. 이를 파라미터를 이용하여 간단히 유성음/무성음/묵음 구간을 분류 할였다. This paper proposes new algorithm for classifying speech signal frame into voiced, unvoiced, silence frame, using the parameters extracted from time domain behavior of speech signal The parameters used in this paper are absolute magnitude, the sum of peaks lager than reference level (T-peak), the ratio of T-peak to absolute magnitude and the magnitude of signal outputs of DPCM. Using this parameters, speech signal is more easily classified into voiced/unvoiced/silence frame.

• 대한음성학회지:말소리
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• 제58호
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• pp.67-81
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• 2006

For a large corpus of time-aligned data, HMM based approaches are most widely used for automatic segmentation, providing a consistent and accurate phone labeling scheme. There are two methods for training in HMM. Flat starting method has a property that human interference is minimized but it has low accuracy. Bootstrap method has a high accuracy, but it has a defect that manual segmentation is required In this paper, a new algorithm is proposed to minimize manual work and to improve the performance of automatic segmentation. At first phase, voiced, unvoiced and silence classification is performed for each speech data frame. At second phase, the phoneme sequence is aligned dynamically to the voiced/unvoiced/silence sequence according to the acoustic phonetic rules. Finally, using these segmented speech data as a bootstrap, phoneme model parameters based on HMM are trained. For the performance test, hand labeled ETRI speech DB was used. The experiment results showed that our algorithm achieved 10% improvement of segmentation accuracy within 20 ms tolerable error range. Especially for the unvoiced consonants, it showed 30% improvement.

• 한국정보통신학회논문지
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• 제18권6호
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• pp.1294-1301
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• 2014

본 논문에서는 화자 인식의 성능을 개선하기 위해서 유성음 및 무성음에 대한 별도의 확률분포 모델링을 사용하였다. 또한, 종래의 멜-주파수 캡스트럼 계수 이외에 유성음 구간에서 추가적으로 왜도, 첨도, 하모닉 대 잡음비 등을 추출하여 활용하였다. 화자 인식을 위한 스코어는 유성음 및 무성음 확률분포 모델에서 각각 구해지는데 전수 조사방식에 의해서 최적의 스코어 결합 가중치가 결정되었다. 제안된 방식의 화자인식기의 성능은 종래의 멜-주파수 캡스트럼 계수 및 화자당 하나의 혼합 가우시안 기반 확률분포 모델링을 사용한 방식과 비교되었으며 실험 결과 제안된 방식이 가우시안 혼합의 수가 낮아질수록 더 큰 성능 향상을 얻음을 알 수 있었다.

• 한국통신학회:학술대회논문집
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• 한국통신학회 1985년도 추계학술발표회 논문집
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• pp.182-185
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• 1985

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 하계학술대회 논문집 G
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• pp.2905-2907
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• 1999

The meaningful speech sound block classification provides very important information in the speech recognition. The following technique of the classification is based on the DWT (discrete wavelet transform), which will provide a more fast algorithm and a useful, compact solution for the pre-processing of speech recognition. The algorithm is implemented to the unvoiced/voiced classification and the denoising.