• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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• pp.207-210
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• 2007

As the Internet is prevalent in our life, harmful contents have been increasing on the Internet, which has become a very serious problem. Among them, pornographic video is harmful as poison to our children. To prevent such an event, there are many filtering systems which are based on the keyword based methods or image based methods. The main purpose of this paper is to devise a system that classifies the pornographic videos based on the audio information. We use Mel-Cepstrum Modulation Energy (MCME) which is modulation energy calculated on the time trajectory of the Mel-Frequency cepstral coefficients (MFCC) and MFCC as the feature vector and Gaussian Mixture Model (GMM) as the classifier. With the experiments, the proposed system classified the 97.5% of pornographic data and 99.5% of non-pornographic data. We expect the proposed method can be used as a component of the more accurate classification system which uses video information and audio information simultaneously.

• 대한음성학회:학술대회논문집
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• 대한음성학회 1996년도 10월 학술대회지
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• pp.333-336
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• 1996

It is widely known that the Japanese alveolar nasal (n) is affected by adjacent vowels in most positions, that is, the variants of the alveolar (n) occur conditionally. The Japanese (n) is palatalized under the influence of vowel (i) or palatal (j). In the articulation of (ni), for instance, the tip and sides of the tongue make wide contact with the palate. It is interesting to know how palatalization occurs and varies during the production in different contexts. In my presentation, the actual realization of the palatalized alveolar nasal in different contexts is examined and clarified by consider me the Electro-palatographic data and examining the articulatory feel ins and auditory impression. As a result, palatalized (equation omitted) occurs either word-initially or inter-vocalically. (equation omitted) in (equation omitted) and (equation omitted) has great palatality. When conditioned by (j), the (equation omitted) in (equation omitted), (equation omitted) and (equation omitted) has full palatality. In each sound the average number of contacted electrodes of the Electro-palatograph at maximum tongue-palate contact is 63 or 100% of the total. To summarize the experimental data, articulatory feel ins and auditory impression, it can be concluded that the (n) followed by or hemmed in (i), (j) is a palatalized nasal (equation omitted).

• 대한음성학회지:말소리
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• 제31_32호
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• pp.223-238
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• 1996

It is widely hewn that the Japanese alveolar nasal [n] is affected by adjacent vowels in most positions, that is, the variants of the alveolar [n] occur conditionally. The Japanese [n] is palatalized under the influence of vowel [i] or palatal [j]. In the articulation of 'に', for instance, the tip and sides of the tongue make wide contact with the palate. It is interesting to know how palatalization occurs and varies during the production in different contexts. In my presentation the actual realization of the palatalized alveolar nasal in different contexts is examined and clarified by considering the Electro-palatographic data and examining the articulatory feeling and auditory impression. As a result, palatalized [${\eta}$] occurs either word-initially- or inter-vocalically. [${\eta}$] in [${\eta}$i] and 'いに'[$i{\eta}$] has great palatality. When conditioned by [j], the [${\eta}$] in 'にゃ'[${\eta}$ja], 'にょ'[${\eta}jo$] and 'にゅ'[${\eta}jw$] has full palatality. In each sound the average number of contacted electrodes of the Electro-palatograph at maximum tongue-palate contact is 63 or 100% of the total. To summarize the experimental data, articulatory feeling and auditory impression, it can be concluded that 'the [n] followed by or hemmed in [i], [j] is a palatalized nasal [${\eta}$].

• 정보교육학회논문지
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• 제2권2호
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• pp.162-173
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• 1998

CAI 프로그램 작성시 자료 공유를 위한 한자 공통코드 정의를 위하여 주요 한자문화권 국가의 한자 코드에 대한 고찰 결과, 한자코드 정의시 발음순과 부수순을 혼용 및 동일 한자에 대한 간체자(중국 GB)와 이체자(일본 Shift JIS)의 개발, 사용이 많아 정체자(KSC 한국)간의 자료 호환시 적절한 변환 장치가 요한다. 발음순 원칙은 동차 이음어가 다른 코드로 인식되어 코드 영역의 낭비가 있고, 부수순 원칙은 자국의 두음법칙에 맞지 않으나 중복 코드를 방지할 수 있어 자료의 복원시 유리하다. KSC 제1수준과 제2수준 한자 영역은 학계 및 업계의 요구를 수용할 수준으로 확대시킬 필요가 있다. 유니코드는 시스템의 호환, 확장에 유리하고, 다양한 문자 표현이 가능하여 교육목적 코드의 일시적인 대안이 될 수 있다.

• 한국음향학회지
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• 제14권4호
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• pp.74-81
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• 1995

본 논문에서는 한국과학기술원(KAIST) 통신연구실에서 개발한 한국어 음성 데이터베이스의 개발에 관하여 기술한다. 음성 데이터베이스의 구축을 위하여 사용된 절차와 환경, 및 데이터베이스의 음성학적, 언어학적 성질들이 상세히 기술된다. 데이터베이스는 음성인식 알고리듬의 개발 및 평가를 위하여 사용되도록 고안되었다. 데이터베이스는 5종류의 음성 데이터, 즉 3천단어 규모의 무역관련 연속음성, 가변길이 연결 숫자음, phoneme-balanced75 고립단어, 지역명 관련 500 고립단어, 한국어 아-세트로 구성되어 있다.

• 전자공학회논문지B
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• 제33B권4호
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• pp.174-182
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• 1996

In continuous speech recognition system, it is possible to implement the system which can handle unlimited number of words by using limited number of phonetic units such as phonemes. Dividing continuous speech into the string of tems of phonemes prior to recognition process can lower the complexity of the system. But because of the coarticulations between neiboring phonemes, it is very difficult ot extract exactly their boundaries. In this paper, we propose the algorithm ot extract short terms which can represent each phonemes instead of extracting their boundaries. The short terms of lower spectral change and higher spectral chang eare detcted. Then phoneme changes are detected using distance measure with this lower spectral change terms, and hgher spectral change terms are regarded as transition terms or short phoneme terms. Finally lower spectral change terms and the mid-term of higher spectral change terms are regarded s the represent each phonemes. The cepstral coefficients and weighted cepstral distance are used for speech feature and measuring the distance because of less computational complexity, and the speech data used in this experimetn was recoreded at silent and ordinary in-dorr environment. Through the experimental results, the proposed algorithm showed higher performance with less computational complexity comparing with the conventional segmetnation algorithms and it can be applied usefully in phoneme-based continuous speech recognition.

• 음성과학
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• 제9권1호
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• pp.137-148
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• 2002

This study investigates phonation modes of vowels following the affricate consonants in Korean and English- -tense affricate /c'/, lenis affricate /c/, and aspirated affricate /$c^{h}$/ for Korean; voiced affricate /$\check{J}$/ and aspirated affricate /c/ for English. The investigation makes significant use of the H1*-H2* measure (a normalized amplitude difference between the first and second harmonics) to provide acoustic correlates of the phonation types. The major findings for English are that the H1*-H2* measure at the vowel onset was significantly larger in post-aspirated position than in post-voiced position. The Korean data showed the H1*-H2* measure at the vowel onset to be significantly higher in the post-aspirated class than in the post-tense class. On the other hand, the Fo values for the post-lenis vowels were significantly lower than those of the other two classes during the first half of the vowel. Based on the phonetic results, this study argues for the need to incorporate the [stiff vocal folds] and [slack vocal folds] features into the phonological treatments of Korean affricates, while maintaining the two features [constricted glottis] and [spread glottis].

• 대한음성학회지:말소리
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• 제65호
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• pp.93-103
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• 2008

In this paper, we propose a method for optimizing a multiple pronunciation dictionary used for modeling pronunciation variations of non-native speech. The proposed method removes some confusable pronunciation variants in the dictionary, resulting in a reduced dictionary size and less decoding time for automatic speech recognition (ASR). To this end, a confusability measure is first defined based on the Levenshtein distance between two different pronunciation variants. Then, the number of phonemes for each pronunciation variant is incorporated into the confusability measure to compensate for ASR errors due to words of a shorter length. We investigate the effect of the proposed method on ASR performance, where Korean is selected as the target language and Korean utterances spoken by Chinese native speakers are considered as non-native speech. It is shown from the experiments that an ASR system using the multiple pronunciation dictionary optimized by the proposed method can provide a relative average word error rate reduction of 6.25%, with 11.67% less ASR decoding time, as compared with that using a multiple pronunciation dictionary without the optimization.

• 전기학회논문지P
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• 제58권3호
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• pp.276-284
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• 2009

In this paper, we develop the image transfer remote maintenance monitoring system for hand held device which can compensate defects of human mistake. The human mistakes always happen when the worker communicate information each other to check and maintenance the equipment of the power plant under bad circumstance such as small place and long distance in power plant. A worker couldn't converse with other when in noisy place like Power plant. So, we make some hand device for handy size and able to converse in noisy place. The developed system can have improvement of productivity through increasing plant operation time. And developed system is composed of advanced H/W(hard ware) system and S/W(soft ware)system. The H/W system consist of media server unit, communication equipment with hand held device, portable camera, mike and head set. The advanced s/w system consist of data base system, client pc(personal computer) real time monitoring system which has server GUI(graphic user interface) program, wireless monitoring program and wire ethernet communication program. The client GUI program is composed of total solution program as pc camera program, and phonetic conversation program etc.. We analyzed the required items and investigated applicable part in the image transfer remote maintenance monitoring system with hand held device. Also we investigated linkage of communication protocol for developed prototype, developed software tool of two-way communication and realtime recording skill of voice with image. We confirmed the efficiency by the field test in preventive maintenance of plant power.

• The Journal of the Acoustical Society of Korea
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• 제25권1E호
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• pp.20-24
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• 2006

In this paper, we focused on the real time implementation of a speech recognition system with medium size of vocabulary considering its application to a mobile phone. First, we developed the PC based variable vocabulary word recognizer having the size of program memory and total acoustic models as small as possible. To reduce the memory size of acoustic models, linear discriminant analysis and phonetic tied mixture were applied in the feature selection process and training HMMs, respectively. In addition, state based Gaussian selection method with the real time cepstral normalization was used for reduction of computational load and robust recognition. Then, we verified the real-time operation of the implemented recognition system on the TMS320C6201 EVM board. The implemented recognition system uses memory size of about 610 kbytes including both program memory and data memory. The recognition rate was 95.86% for ETRI 445DB, and 96.4%, 97.92%, 87.04% for three kinds of name databases collected through the mobile phones.