• 말소리와 음성과학
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• 제5권3호
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• pp.73-81
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• 2013

Coda in the word-medial position plays an important role in acquisition of our speech. Accuracy of the coda in the word-medial position is important as a diagnostic indicator since it has a close relationship with degrees of disorder. Coda in the word-medial position only appears in condition of connecting two vowels and the sequence causes diverse phonological processes to happen. The coda in the word-medial position differs in production difficulty by the initial sound in the sequence. Accordingly, this study aims to examine the tendency of producing a coda in the word-medial position with consideration of an optional phonological process in spontaneous speech of three and four year old children. Data was collected from 24 children (four groups by age) without speech and language delay. The results of the study are as follows: 1) Sonorant coda in the word-medial position showed a high production frequency in manner of articulation, and alveolar in place of articulation. When the coda in the word-medial position is connected to an initial sound in the same place of articulation, it revealed a high frequency of production. 2) The coda in word-medial position followed by an initial alveolar stop revealed a high error rate. Error patterns showed regressive assimilation predominantly. 3) The order of difficulty that Children had producing codas in the word-medial position was $/k^{\neg}/$, $/p^{\neg}/$, /m/, /n/, /ŋ/ and /l/. Those results suggest that in targeting coda in the word-medial position for evaluation, we should consider optional phonological process as well as the following initial sound. Further studies would be necessary which codas in the word-medial position will be used for therapeutic purpose.

• 제어로봇시스템학회논문지
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• 제16권12호
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• pp.1137-1142
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• 2010

This paper studied the speech parameters less affected by the human emotion for the development of the robust speech recognition system. For this purpose, the effect of emotion on the speech recognition system and robust speech parameters of speech recognition system were studied using speech database containing various emotions. In this study, mel-cepstral coefficient, delta-cepstral coefficient, RASTA mel-cepstral coefficient and frequency warped mel-cepstral coefficient were used as feature parameters. And CMS (Cepstral Mean Subtraction) method were used as a signal bias removal technique. Experimental results showed that the HMM based speaker independent word recognizer using vocal tract length normalized mel-cepstral coefficient, its derivatives and CMS as a signal bias removal showed the best performance of 0.78% word error rate. This corresponds to about a 50% word error reduction as compare to the performance of baseline system using mel-cepstral coefficient, its derivatives and CMS.

• 전자공학회논문지B
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• 제30B권2호
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• pp.51-58
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• 1993

This paper describes the development of a speaker-dependent isolated word recognizer as applied to voice dialing in a car noise environment. for this purpose, several methods to improve performance under such condition are evaluated using database collected in a small car moving at 100km/h The main features of the recognizer are as follow: The endpoint detection error can be reduced by using the magnitude of the signal which is inverse filtered by the AR model of the background noise, and it can be compensated by using variants of the DTW algorithm. To remove the noise, an autocorrelation subtraction method is used with the constraint that residual energy obtainable by linear predictive analysis should be positive. By using the noise rubust distance measure, distortion of the feature vector is minimized. The speech recognizer is implemented using the Motorola DSP56001(24-bit general purpose digital signal processor). The recognition database is composed of 50 Korean names spoken by 3 male speakers. The recognition error rate of the system is reduced to 4.3% using a single reference pattern for each word and 1.5% using 2 reference patterns for each word.

• 음성과학
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• 제7권2호
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• pp.193-208
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• 2000

This paper describes a multi-step method of intonation modeling for corpus-based Korean speech synthesizer. We selected 1833 sentences considering various syntactic structures and built a corresponding speech corpus uttered by a female announcer. We detected the pitch using laryngograph signals and manually marked the prosodic boundaries on recorded speech, and carried out the tagging of part-of-speech and syntactic analysis on the text. The detected pitch was separated into 3 frequency bands of low, mid, high frequency components which correspond to the baseline, the word tone, and the syllable tone. We predicted them using the CART method and the Viterbi search algorithm with a word-tone-dictionary. In the collected spoken sentences, 1500 sentences were trained and 333 sentences were tested. In the layer of word tone modeling, we compared two methods. One is to predict the word tone corresponding to the mid-frequency components directly and the other is to predict it by multiplying the ratio of the word tone to the baseline by the baseline. The former method resulted in a mean error of 12.37 Hz and the latter in one of 12.41 Hz, similar to each other. In the layer of syllable tone modeling, it resulted in a mean error rate less than 8.3% comparing with the mean pitch, 193.56 Hz of the announcer, so its performance was relatively good.

• 말소리와 음성과학
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• 제15권2호
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• pp.13-20
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• 2023

This study explores the potential of automated speech recognition (ASR) in assessing English learners' pronunciation. We employed ASR technology, acknowledged for its impartiality and consistent results, to analyze speech audio files, including synthesized speech, both native-like English and Korean-accented English, and speech recordings from a native English speaker. Through this analysis, we establish baseline values for the word error rate (WER). These were then compared with those obtained for human raters in perception experiments that assessed the speech productions of 30 first-year college students before and after taking a pronunciation course. Our sub-group analyses revealed positive training effects for Whisper, an ASR tool, and human raters, and identified distinct human rater strategies in different assessment aspects, such as proficiency, intelligibility, accuracy, and comprehensibility, that were not observed in ASR. Despite such challenges as recognizing accented speech traits, our findings suggest that digital tools such as ASR can streamline the pronunciation assessment process. With ongoing advancements in ASR technology, its potential as not only an assessment aid but also a self-directed learning tool for pronunciation feedback merits further exploration.

• 한국음향학회지
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• 제16권2호
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• pp.49-57
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• 1997

본 논문에서는 대어휘 연속음성인식을 위한 확률 발음사전 모델에 대해서 제안하였다. 확률 발음 사전은 HMM과 같이 단위음소 상태의 Markov chain으로 이루어져 있으며, 각 음소 상태들은 음소들에 대한 확률 분포 함수로 표현된다. 확률 발음 사전의 생성은 음성자료와 음소 모델을 이용하여 음소 단위의 분할과 인식을 통해서 자동으로 생성되게 된다. 제안된 확률 발음 사전은 단어내 변이와 단어간 변이를 모두 효과적으로 표현할 수 있었으며, 인식 모델과 인식기의 특성을 반영함으로써 전체 인식 시스템의 성능을 보다 높일 수 있었다. 3000 단어 연속음성인식 실험 결과 확률 발음 사전을 사용함으로써 표준 발음 표기를 사용하는 인식 시스템에 비해 단어 오류율은 23.6%, 문장 오류율은 10% 정도를 감소시킬 수 있었다.

• 한국정보통신학회논문지
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• 제20권3호
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• pp.464-470
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• 2016

본 논문에서는 음성의 특성 지표를 이용한 음성 인식용 데이터베이스 검증 시스템의 개발 내용을 소개하고 이 시스템의 핵심 기술인 음성 특성 지표 추출 알고리즘을 설명한다. 선행 연구에서는 본 시스템에 필요한 효과적인 음성 인식 성능 지표를 생성하기 위해 대표적인 음성 인식 성능 지표인 단어 오인식률(Word Error Rate, WER)과 상관도가 높은 여러 가지 음성 특성 지표들을 조합하여 새로운 성능 지표를 생성하였다. 생성된 음성 인식 성능 지표는 다양한 잡음 환경에서 각 음성 특성 지표를 단독으로 사용할 때보다 단어 오인식률과 높은 상관도를 나타내어 음성 인식 성능을 예측하는데 효과적임을 입증 하였다. 본 실험에서는 선행 연구에서 조합에 사용한 이차적인 음성 인식기에서 추출된 음향 모델 확률 값을 GMM(Gaussian Mixture Model) 음향 모델 확률 값으로 대체해 조합함으로써 시스템 구축 시 다른 음성 인식기에 대한 의존성을 감소시킨다.

• ETRI Journal
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• 제22권3호
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• pp.10-18
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• 2000

Although an n-th order cross-entropy (nCE) error function resolves the incorrect saturation problem of conventional error backpropagation (EBP) algorithm, performance of multilayer perceptrons (MLPs) trained using the nCE function depends heavily on the order of nCE. In this paper, we propose an adaptive learning rate to markedly reduce the sensitivity of MLP performance to the order of nCE. Additionally, we propose to limit error signal values at out-put nodes for stable learning with the adaptive learning rate. Through simulations of handwritten digit recognition and isolated-word recognition tasks, it was verified that the proposed method successfully reduced the performance dependency of MLPs on the nCE order while maintaining advantages of the nCE function.

• 대한한의학원전학회지
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• 제32권3호
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• pp.47-57
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• 2019

Objectives : We aim to assist in choosing an appropriate method for word extraction when analyzing East Asian Traditional Medical texts based on unsupervised learning. Methods : In order to assign ranks to substrings, we conducted a test using one method(BE:Branching Entropy) for exterior boundary value, three methods(CS:cohesion score, TS:t-score, SL:simple-ll) for interior boundary value, and six methods(BExSL, BExTS, BExCS, CSxTS, CSxSL, TSxSL) from combining them. Results : When Miss Rate(MR) was used as the criterion, the error was minimal when the TS and SL were used together, while the error was maximum when CS was used alone. When number of segmented texts was applied as weight value, the results were the best in the case of SL, and the worst in the case of BE alone. Conclusions : Unsupervised-Learning-Based Word Extraction is a method that can be used to analyze texts without a prepared set of vocabulary data. When using this method, SL or the combination of SL and TS could be considered primarily.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1988년도 전기.전자공학 학술대회 논문집
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• pp.194-197
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• 1988

The unequal bit-error-probability of rate r=b/n binary convolutional code is analyzed. The error protection affored each digit of the b-tuple information word can be different from that afforded other digit. The property of the unequal protection can be applied to transmitting sampled data in PCM system.