• 제목/요약/키워드: phonetic information

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계산주의적 모델을 이용한 한국어 시각단어 재인에서 나타나는 이웃효과 (The Neighborhood Effects in Korean Word Recognition Using Computation Model)

  • 박기남;권유안;임희석;남기춘
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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    • pp.295-297
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    • 2007
  • This study suggests a computational model to inquire the roles of phonological information and orthography information in the process of visual word recognition among the courses of language information processing and the representation types of the mental lexicon. As the result of the study, the computational model showed the phonological and orthographic neighborhood effect among language phenomena which are shown in Korean word recognition, and showed proofs which implies that the mental lexicon is represented as phonological information in the process of Korean word recognition.

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Landmark-Guided Segmental Speech Decoding for Continuous Mandarin Speech Recognition

  • Chao, Hao;Song, Cheng
    • Journal of Information Processing Systems
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    • 제12권3호
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    • pp.410-421
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    • 2016
  • In this paper, we propose a framework that attempts to incorporate landmarks into a segment-based Mandarin speech recognition system. In this method, landmarks provide boundary information and phonetic class information, and the information is used to direct the decoding process. To prove the validity of this method, two kinds of landmarks that can be reliably detected are used to direct the decoding process of a segment model (SM) based Mandarin LVCSR (large vocabulary continuous speech recognition) system. The results of our experiment show that about 30% decoding time can be saved without an obvious decrease in recognition accuracy. Thus, the potential of our method is demonstrated.

지능형 로봇 '웨버'를 위한 음원 추적 기술 (Sound Localization Technique for Intelligent Service Robot 'WEVER')

  • 이지연;한민수;지수영;조영조
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2005년도 추계 학술대회 발표논문집
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    • pp.117-120
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    • 2005
  • This paper suggests an algorithm that can estimate the direction of the sound source in realtime. Our intelligent service robot, WEVER, is used to implement the proposed method at the home environment. The algorithm uses the time difference and sound intensity information among the recorded sound source by four microphones. Also, to deal with noise of robot itself, the kalman filter is implemented. The proposed method takes shorter execution time than that of an existing algorithm to fit the real-time service robot. The result shows relatively small error within the range of ${\pm}$ 7 degree.

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억양과 초점에 관한 화용론적 연구 (A pragmatically-oriented study of intonation and focus)

  • 이영길
    • 대한음성학회지:말소리
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    • 제38호
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    • pp.1-24
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    • 1999
  • There is an indisputable connection between prosody and focus. The focal prominence in Korean, a prosodic realization of pitch prominence in an utterance, defines a focused constituent, the domain of which is identified by the Focus Identification Principle. To this is added the Basic Focus Rule which makes it possible to capture and interpret the focal domain, which can then be tested against the available context. The focal domain can be contextually made available by setting it off with information structure boundaries(I/S) identified by the Information Structure Identification Principle. The fragment of the utterance enclosed within the IS boundaries can be recognized as 'new' information with the help of the Focus Domain Identification Rule. Since information structures are pragmatically tied to semantic levels of grammatical systems, the Basic Focus Rule is now replaced by the Focal Prominence Principle ensuring the focal prominence within the focal domain. Close relationships exist between patterns of intonation and their expressiveness in terms of giving a pragmatically-oriented description of focus. This is particularly manifested in Korean sentences containing contrastiveness.

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상태의 고유시간 정보를 포함하는 Hidden Markov Model (Hidden Markov Models Containing Durational Information of States)

  • 조정호;홍재근;김수중
    • 대한전자공학회논문지
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    • 제27권4호
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    • pp.636-644
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    • 1990
  • Hidden Markov models(HMM's) have been known to be useful representation for speech signal and are used in a wide variety of speech systems. For speech recognition applications, it is desirable to incorporate durational information of states in model which correspond to phonetic duration of speech segments. In this paper we propose duration-dependent HMM's that include durational information of states appropriately for the left-to-right model. Reestimation formulae for the parameters of the proposed model are derived and their convergence is verified. Finally, the performance of the proposed models is verified by applying to an isolated word, speaker independent speech recognition system.

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SiTEC의 공동 이용을 위한 음성 코퍼스 구축 현황 및 계획 (Current States and Future Plans at SiTEC for Speech Corpora for Common Use)

  • 김봉완;최대림;김영일;이광현;이용주
    • 대한음성학회지:말소리
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    • 제46호
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    • pp.175-185
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    • 2003
  • To support speech information technology industry it is vital to create and distribute standardized speech corpora to be used for the development of products and technologies. In this article we introduce speech corpora created by Speech Information Technology & Industry Promotion Center(SiTEC) during its 1st and 2nd fiscal years (2001/5/1-2003/4/30) and plans for those corpora which is being created currently or will be created in near future. We introduce the corpus for car application to expand speech information technology to the field of traditional industry, the corpora for foreign languages to support exportation, the corpus for basic research for the sake of application in the industry, the corpora for common use, and others.

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적응 콤 필터링을 이용한 이동 통신 환경에서의 강인한 음성 인식 (Robust Speech Recognition using Adaptive Comb Filtering in Mobile Communication Environment)

  • 박정식;정규준;오영환
    • 대한음성학회지:말소리
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    • 제46호
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    • pp.65-76
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    • 2003
  • In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well-known method for noise reduction, but requires correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.

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위장발화의 단모음 포만트 연구 (A Study on the Vowel Fomants in Disguised Speech)

  • 노석은;박미경;조민하;신지영;강선미
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2004년도 춘계 학술대회 발표논문집
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    • pp.215-218
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    • 2004
  • The aim of this paper is to analyze the acoustic features for disguised voice. In this paper we examined the features such as pitch range, vowel formants(F1, F2, F3, F4). So the result of the analysis is as follows. : (1) Pitch range and average of pitch value is very important cue for speaker verification. (2) F3-F2 is also important cue for speaker verification (3) /a/ is more verified than other vowels.

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외국어 발화오류 검출 음성인식기를 위한 스코어링 기법 (Machine scoring method for speech recognizer detection mispronunciation of foreign language)

  • 강효원;배민영;이재강;권철홍
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2004년도 춘계 학술대회 발표논문집
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    • pp.239-242
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    • 2004
  • An automatic pronunciation correction system provides users with correction guidelines for each pronunciation error. For this purpose, we propose a speech recognition system which automatically classifies pronunciation errors when Koreans speak a foreign language. In this paper, we also propose machine scoring methods for automatic assessment of pronunciation quality by the speech recognizer. Scores obtained from an expert human listener are used as the reference to evaluate the different machine scores and to provide targets when training some of algorithms. We use a log-likelihood score and a normalized log-likelihood score as machine scoring methods. Experimental results show that the normalized log-likelihood score had higher correlation with human scores than that obtained using the log-likelihood score.

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Detection of Pathological Voice Using Linear Discriminant Analysis

  • Lee, Ji-Yeoun;Jeong, Sang-Bae;Choi, Hong-Shik;Hahn, Min-Soo
    • 대한음성학회지:말소리
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    • 제64호
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    • pp.77-88
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    • 2007
  • Nowadays, mel-frequency cesptral coefficients (MFCCs) and Gaussian mixture models (GMMs) are used for the pathological voice detection. This paper suggests a method to improve the performance of the pathological/normal voice classification based on the MFCC-based GMM. We analyze the characteristics of the mel frequency-based filterbank energies using the fisher discriminant ratio (FDR). And the feature vectors through the linear discriminant analysis (LDA) transformation of the filterbank energies (FBE) and the MFCCs are implemented. An accuracy is measured by the GMM classifier. This paper shows that the FBE LDA-based GMM is a sufficiently distinct method for the pathological/normal voice classification, with a 96.6% classification performance rate. The proposed method shows better performance than the MFCC-based GMM with noticeable improvement of 54.05% in terms of error reduction.

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