• Title/Summary/Keyword: Phoneme

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A Study on Construction of Acoustical Phoneme Models Using Hidden Markov Network (Hidden Markov Network를 이용한 음향학적 음소모델 작성에 관한 검토)

  • Oh Se-Jin;Lim Young-Choon;Hwang Cheol-Jun;Kim Bum-Koog;Chung Hyun-Yeol
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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    • pp.29-32
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    • 2000
  • 본 논문에서는 음성인식 시스템의 음향모델 개선을 위한 기초적 연구로서, 문맥적인 요소를 필요로 하는 SSS(Successive State Splitting)와 필요로 하지 않는 SSS-free 알고리즘을 이용한 HMnet(Hidden Markov Network) 음향모델 작성방법에 대해 검토하고 작성한 음향모델을 한국어에 적용하여 그 유효성을 확인하였다. HMnet을 이용한 음소모델의 작성방법은 전체 학습 데이터에 대해서 각각 2개의 상태를 가지는 초기 모델을 작성한 후, 이를 시간과 문맥방향으로의 최대 분포를 가지는 상태를 재분할한 후 임의의 상태수가 될 때까지 상태분할을 계속적으로 수행케 하여 각 음소모델을 작성하게 된다. 작성한 HMnet 음향모델의 유효성을 확인하기 위해 ETRI 445 단어의 3인에 대한 화자종속 음소인식 실험을 수행하였다. 인식실험 결과, SSS 알고리즘을 이용한 화자종속실험의 경우 상태수 520에서 평균 $62.8\%$의 인식률을, SSS-free 알고리즘의 경우 상태수 420에서 평균 $64.2\%$의 인식률을 얻었다. 이 결과는 HMM을 이용한 경우(약$43.4\%$)보다 $20\%$이상의 인식률 향상을 보여 이 알고리즘의 유효성을 확인할 수 있었다. SSS와 SSS-free를 비교한 경우, SSS-free가 SSS보다 낮은 상태수에서 평균 $1.4\% 향상된 인식률을 보였다.

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Pitch Period Detection Algorithm Using Rotation Transform of AMDF (AMDF의 회전변환을 이용한 피치 주기 검출 알고리즘)

  • Seo, Hyun-Soo;Bae, Sang-Bum;Kim, Nam-Ho
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • v.9 no.2
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    • pp.1019-1022
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    • 2005
  • As recent information communication technology is rapidly developed, a lot of researches related to speech signal processing have been processed. So pitch period is applied as important factor to many application fields such as speech recognition, speaker identification, speech analysis and synthesis. Therefore, many algorithms related to pitch detection have been proposed in time domain and frequency domain and AMDF(average magnitude difference function) which is one of pitch detection algorithms in time domain chooses time interval from valley to valley as pitch period. But, in selection of valley point to detect pitch period, complexity of the algorithm is increased. So in this paper we proposed pitch detection algorithm using rotation transform of AMDF, that taking the global minimum valley point as pitch period and established a threshold about the phoneme in beginning portion, to exclude pitch period selection. and compared existing methods with proposed method through simulation.

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An On-line Speech and Character Combined Recognition System for Multimodal Interfaces (멀티모달 인터페이스를 위한 음성 및 문자 공용 인식시스템의 구현)

  • 석수영;김민정;김광수;정호열;정현열
    • Journal of Korea Multimedia Society
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    • v.6 no.2
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    • pp.216-223
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    • 2003
  • In this paper, we present SCCRS(Speech and Character Combined Recognition System) for speaker /writer independent. on-line multimodal interfaces. In general, it has been known that the CHMM(Continuous Hidden Markov Mode] ) is very useful method for speech recognition and on-line character recognition, respectively. In the proposed method, the same CHMM is applied to both speech and character recognition, so as to construct a combined system. For such a purpose, 115 CHMM having 3 states and 9 transitions are constructed using MLE(Maximum Likelihood Estimation) algorithm. Different features are extracted for speech and character recognition: MFCC(Mel Frequency Cepstrum Coefficient) Is used for speech in the preprocessing, while position parameter is utilized for cursive character At recognition step, the proposed SCCRS employs OPDP (One Pass Dynamic Programming), so as to be a practical combined recognition system. Experimental results show that the recognition rates for voice phoneme, voice word, cursive character grapheme, and cursive character word are 51.65%, 88.6%, 85.3%, and 85.6%, respectively, when not using any language models. It demonstrates the efficiency of the proposed system.

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Improvement of Synthetic Speech Quality using a New Spectral Smoothing Technique (새로운 스펙트럼 완만화에 의한 합성 음질 개선)

  • 장효종;최형일
    • Journal of KIISE:Software and Applications
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    • v.30 no.11
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    • pp.1037-1043
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    • 2003
  • This paper describes a speech synthesis technique using a diphone as an unit phoneme. Speech synthesis is basically accomplished by concatenating unit phonemes, and it's major problem is discontinuity at the connection part between unit phonemes. To solve this problem, this paper proposes a new spectral smoothing technique which reflects not only formant trajectories but also distribution characteristics of spectrum and human's acoustic characteristics. That is, the proposed technique decides the quantity and extent of smoothing by considering human's acoustic characteristics at the connection part of unit phonemes, and then performs spectral smoothing using weights calculated along a time axis at the border of two diphones. The proposed technique reduces the discontinuity and minimizes the distortion which is caused by spectral smoothing. For the purpose of performance evaluation, we tested on five hundred diphones which are extracted from twenty sentences using ETRI Voice DB samples and individually self-recorded samples.

Comparison of Performance on Superordinate Word Tasks in Elderly and Young Adults (노년층과 청년층의 상위범주어 과제 수행력 비교)

  • Kim, Hyung Moo;Yoon, Ji Hye
    • 재활복지
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    • v.20 no.4
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    • pp.229-246
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    • 2016
  • The aim of this study is to conduct superordinate word selection task to compare their performance and reaction time, and superordinate word writing task to compare the differences in their performance and error pattern in 40 elderly adults and 43 young adults. As a result, first, in both tasks, elderly adults had a smaller number of correct responses. Second, elderly adults showed slower reaction time than young adults. Third, in superordinate word writing task, elderly adults showed more relevant errors than irrelevant errors. The reason elderly adults had a smaller number of correct responses in both tasks was that the links among the pieces of information in the semantic lexicon weakened or deteriorated due to normal aging. Slower reaction time was based on neurophysiological changes of the brain and cognitive processing speed. In addition, the relevant errors showed that they could access the lexicon for target words and produce explanation the relevant characteristics, even though they could not retrieve the target words.

Changes of Speech Discrimination Score Depending on Inter-syllable Pause Duration in Normal Hearing Children (정상 청력 아동의 음절 간 쉼 간격에 따른 어음이해도 변화)

  • Park, J.I.;Lee, J.Y.;Heo, S.D.
    • Journal of rehabilitation welfare engineering & assistive technology
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    • v.8 no.2
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    • pp.139-144
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    • 2014
  • Speech discrimination is affected by the speed of speech. The speed of speech can be adjusted at the pause duration, the pause duration can take the resting time to avoid in overloading information. The study will be examine the effects of aging and audiological rehabilitation, and the auditory processing as basic research to investigate the normative data. 7 boys and 8 girls were participated. They have no problem with speech language pathologically and audiologically. There are 4 sets of test implement, and each test set was made out with 20 3-syllable words. Pause duration of all of these words are adjusted in normal(250 ms), slow(500 ms) and very slow(1000 ms). There are 4 words for a multiple-choice that including one word with written correctly and three words with written 1 phoneme wrong. Participant hear the word, and then have to choose one. Speech discrimination score in 250, 500, 1,000 ms of pause duration were $73{\pm}19.4%$, $84{\pm}12.2%$, $88{\pm}8.8%$, respectively.

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Analysis of Error Characteristics and Usabilities for Korean Consonant Perception Test (한국자음지각검사의 오류특성 및 유용성 분석)

  • Kim, Dong Chang;Kim, Jin Sook;Lee, Kyoung Won
    • 재활복지
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    • v.18 no.4
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    • pp.295-314
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    • 2014
  • The purpose of this study was to supply the baseline data for auditory rehabilitation in the field through error type and rate of the phoneme which the hearing impaired feel difficulty to discriminate. Thirty participants with sensorineural hearing loss heard KCPT lists through recorded voice by male and female to get the data about error type and KCPT score accordance with talker's gender. In the initial consonant test list, /ㄷ/, /ㅂ/, /ㅃ/, /ㅉ/, /ㅌ/ showed more than 30% error rate while /ㄱ/and /ㄷ/ showed in final consonant test list. The most common error type was the initial consonant substitution or the final consonant substitution for the initial or final consonant test lists. Talker's gender effect was not signigicant showing no statistical difference between the scores when compared results from male voice and female voice. It means that KCPT can be used regardless of talker's gender in clinics.

A Study on the Characteristics of Errors Type for Wellness of Alzheimer's Dementia Patients in the Naming Task (알츠하이머성 치매환자의 웰니스를 위한 명명하기 과제에서의 오류유형 특성 연구)

  • Kang, Min-Gu
    • Journal of Korea Entertainment Industry Association
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    • v.14 no.8
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    • pp.213-219
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    • 2020
  • The purpose of this study was to investigate the characteristics of error types in naming task for 8 questionable demeatia groups, 9 definite dementia groups, and 10 normal groups. The items of naming error analysis were classified into visual perception errors, semantic association errors, semantic non-correlation errors, phoneme errors, Don't Know, and No Response. For the analysis, descriptive statistics analysis, analysis of variance, and multivariate analysis of variance were conducted using SPSS 21.0. As a result, there was a significant difference in the error rate between groups according to the error type. The errors that showed significant differences between the normal group and the other two groups were visual perception errors and semantic non-related errors. The error of non-response was different from the dementia confirmation group, but there was no significant difference from the dementia suspicion group. These results showed that Alzheimer's patients had a defect in confrontation naming ability. Also, it was found that it is appropriate to provid other clues when the defects caused by the degeneration of a specific step during the information processing process become severe.

Web strain based prediction of web distortion influence on the elastic LTB limiting length

  • Bas, Selcuk
    • Steel and Composite Structures
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    • v.43 no.2
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    • pp.271-278
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    • 2022
  • Buckling is one of the most critical phoneme in the design of steel structures. Lateral torsional buckling (LTB) is particularly significant for slender beams generally subjected to loading in plane. The web distortion effects on LTB are not addressed explicitly in standards for flexural design of steel I-section members. Hence, the present study is focused to predict the influence of the web distortion on the elastic (Lr) limiting lengths given in American Institute of Steel Construction (AISC) code for the lateral torsional buckling (LTB) behavior of steel beams due to no provision in the code for consideration of web distortion. For this aim, the W44x335 beam is adopted in the buckling analysis carried out by the ABAQUS finite element (FE) program since it is one of the most critical sections in terms of lateral torsional buckling (LTB). The strain results at mid-height of the web at mid-span of the beam are taken into account as the monitoring parameters. The web strain results are found to be relatively greater than the yield strain value when L/Lr is equal to 1.0. In other words, the ratio of L/Lr is estimated from the numerical analysis to be about 1.5 when the beam reaches its first yielding at mid-span of the beam at mid-height of the section. Due to the effect of web distortion, the elastic limiting length (Lr) from the numerical analysis is obtained to be considered as greater than the calculated length from the code formulation. It is suggested that the formulations of the limiting length proposed in the code can be corrected considering the influence of the web distortion. This correction can be a modification factor or a shape factor that reduces sectional slenderness for the LTB formulation in the code.

Improvements on Speech Recognition for Fast Speech (고속 발화음에 대한 음성 인식 향상)

  • Lee Ki-Seung
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.2
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    • pp.88-95
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    • 2006
  • In this Paper. a method for improving the performance of automatic speech recognition (ASR) system for conversational speech is proposed. which mainly focuses on increasing the robustness against the rapidly speaking utterances. The proposed method doesn't require an additional speech recognition task to represent speaking rate quantitatively. Energy distribution for special bands is employed to detect the vowel regions, the number of vowels Per unit second is then computed as speaking rate. To improve the Performance for fast speech. in the pervious methods. a sequence of the feature vectors is expanded by a given scaling factor, which is computed by a ratio between the standard phoneme duration and the measured one. However, in the method proposed herein. utterances are classified by their speaking rates. and the scaling factor is determined individually for each class. In this procedure, a maximum likelihood criterion is employed. By the results from the ASR experiments devised for the 10-digits mobile phone number. it is confirmed that the overall error rate was reduced by $17.8\%$ when the proposed method is employed