• 한국컴퓨터정보학회논문지
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• 제5권4호
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• pp.48-54
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• 2000

본 논문에서는 한ㆍ영 기계번역을 위한 한국어의 품사를 분류하였고 각 품사의 형태론적 특징을 고찰하였다. 한국어 표준문법에서 제시되는 품사 분류 기준은 의미, 기능, 형태의 세 가지 기준을 적용하고 있으며, 자연언어처리에서도 같은 분류 기준을 바탕으로 하고 있다. 품사 분류에 여러 가지 기준을 적용하는 것은 문법구조 이해 및 품사 분류를 어렵게 한다. 또한 한 영 기계번역시 품사의 불일치로 전처리가 필요하다. 이러한 문제를 해결하기 위하여 본 논문에서는 하나의 기준을 적용하여 품사를 분류하였다. 방법으로 한국어 표준문법에 의하여 말뭉치에 태깅하고 문제점을 찾아내며, 새로운 기준에 의하여 품사를 분류하였다. 본 논문에서 분류된 품사는 한국어 문장에서 통사적 역할이 동일하고, 영어에서의 사전 품사와 동일하며, 품사 분류의 모호성을 제거하고, 한국어의 문장 구조를 명확히 표현한다. 또한 한ㆍ영 기계번역시 패턴 매칭에 의한 목적언어 생성이 가능하게 한다.

• International Journal of Computer Science & Network Security
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• 제22권4호
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• pp.245-253
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• 2022

Commenced in 1954 by IBM, machine translation has expanded immensely, particularly in this period. Machine translation can be broken into seven main steps namely- token generation, analyzing morphology, lexeme, tagging Part of Speech, chunking, parsing, and disambiguation in words. Morphological analysis plays a major role when translating Indian languages to develop accurate parts of speech taggers and word sense. The paper presents various machine translation methods used by different researchers for Indian languages along with their performance and drawbacks. Further, the paper concentrates on parts of speech (POS) tagging in Marathi dialect using various methods such as rule-based tagging, unigram, bigram, and more. After careful study, it is concluded that for machine translation, parts of speech tagging is a major step. Also, for the Marathi language, the Hidden Markov Model gives the best results for parts of speech tagging with an accuracy of 93% which can be further improved according to the dataset.

• 말소리와 음성과학
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• 제11권4호
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• pp.1-8
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• 2019

This study aims to investigate how prosodic prominence is perceived in relation to word class information (or parts-of-speech) by Korean learners of English compared with native English speakers in public speech. Two groups, Korean learners of English and native English speakers, were asked to judge words perceived as prominent simultaneously while listening to a speech. Parts-of-speech and three acoustic cues (i.e., max F0, mean phone duration, and mean intensity) were analyzed for each word in the speech. The results showed that content words tended to be higher in pitch and longer in duration than function words. Both groups of listeners rated prominence on content words more frequently than on function words. This tendency, however, was significantly greater for Korean learners of English than for native English speakers. Among the parts-of-speech of the content words, Korean learners of English were more likely than native English speakers to judge nouns and verbs as prominent. This study presents evidence that Korean learners of English consider most, if not all, content words as landing locations of prosodic prominence, in alignment with the previous study on the production of prominence.

• Journal of information and communication convergence engineering
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• 제1권4호
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• pp.209-212
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• 2003

In this paper, we proposed new speech feature parameter through parts-based feature extraction of speech spectrum using Non-Negative Matrix Factorization (NMF). NMF can effectively reduce dimension for multi-dimensional data through matrix factorization under the non-negativity constraints, and dimensionally reduced data should be presented parts-based features of input data. For speech feature extraction, we applied Mel-scaled filter bank outputs to inputs of NMF, than used outputs of NMF for inputs of speech recognizer. From recognition experiment result, we could confirm that proposed feature parameter is superior in recognition performance than mel frequency cepstral coefficient (MFCC) that is used generally.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1991년도 학술발표회 논문집
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• pp.7-10
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• 1991

In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

• The Journal of the Acoustical Society of Korea
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• 제25권4E호
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• pp.142-147
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• 2006

Blind signal separation is known as a powerful tool for enhancing noisy speech in many real world environments. In this paper, it is demonstrated that the performance of blind signal separation can be further improved by combining with a null beamformer (NBF). Cascading the blind source separation with null beamforming is equivalent to the decomposition of the received signals into the direct parts and reverberant parts. Investigation of beam patterns of the null beamformer and blind signal separation reveals that directional null of NBF reduces mainly direct parts of the unwanted signals whereas blind signal separation reduces reverberant parts. Further, it is shown that the decomposition of received signals can be exploited to solve the local stability problem. Therefore, faster and improved separation can be obtained by removing the direct parts first by null beamforming. Simulation results using real office recordings confirm the expectation.

• 대한음성학회지:말소리
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• 제47호
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• pp.141-151
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• 2003

The speech intelligibility of congenitally deafened children shows the change after cochlear implantation. The predicting factors of change in speech intelligibility are the age of implantation, the duration of implant use, and communication mode etc.. Among these factors, the age of implantation seems to be one of the most important predictors. But those factors including age of implantation can explain only some parts of the variance. Therefore, the further study to find the factors which affect the speech intelligibility should be done.

• 전자공학회논문지C
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• 제35C권7호
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• pp.59-68
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• 1998

This paper is to develop an algorithm for the unlimited korean speech synthesis. We present the results controlled of prosodic factors with isolated words as aynthesis basis unit int he time domain. With a new pitch-synchronous and parametric speech synthesis mehtod in the time domain here we mainly present the results of controlled prosody factors such a spitch periods, energy envelops and durations and the evaluaton of synthetic speech qualities. In the case of synthesis, it is possible ot synthesize connected words by controlling of a continuous unified prosody that makes to improve the naturalities. In the results of experiment, it also has been to be improved uncontinuities of pitch and zeroing of energy in the junction parts of speech waveforms. Specially it has been to be possible to synthesize speeches with unlimitted durations and tones. So on it makes the noisiness and the clearness better by improving the degradation effects from the phase distortion due to the discontinuities in the waveform connection parts.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1991년도 학술발표회 논문집
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• pp.15-20
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• 1991

In a text-to-speech system, sound units (e. q., phonemes, words, or phrases) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is availalbe for joining the segment together.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1998년도 추계학술대회
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• pp.163-164
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• 1998

In this paper the digital speech processing part of cochlear implant for sensorineural disorderly patients is implemented and simulated. We implement the speech processing part by dividing into three small parts - Filterbank, Pitch Detect, and Bandmapping parts. With the result, we conclude digital speech processing algorithm is implemented in FPGA perfectly. This means that cochlear implant can be made very small size.