• Journal of The Korean Association of Information Education
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• v.6 no.3
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• pp.308-316
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• 2002

In this paper, a learning method of Pansori using voice bulletin board system is introduced. Also, the influence of the proposed approach on learners is analyzed by qualitative methods such as participatory observation, interview and video recording. The results of the qualitative analysis shows that the proposed approach can be effectively applied to Pansori learning. Futhermore, it can be seen that learners have a great interest in Korean traditional folk music.

• Proceedings of the Korea Information Processing Society Conference
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• 2012.04a
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• pp.908-911
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• 2012

인간의 본능적인 욕구 중 생리적 욕구는 생존을 위해서 불가결한 것 중 하나이며 이러한 생리적 요구엔 성욕이 포함되어 있다. 성욕은 외부자극으로 인하여 욕구가 충동되며 도파민과 테스토스테론의 호르몬 분비가 일어나 성적 충동을 증가시켜 신체변화에 영향을 미친다. 따라서 본 논문에서는 성욕을 자극하여 성적 충동이 증가되었을 때 목소리의 변화를 분석하는 연구를 수행하였다. 이를 위해 성적 충동이 증가되기 전과 후의 음성을 수집하고 성대 관련 음성분석 요소인 Pitch, Intensity 기술을 적용하여 변화된 음성의 특징을 추출하였다.

• Proceedings of the KSLP Conference
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• 1996.11a
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• pp.87-87
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• 1996

객관적인 검사도구인 Dr. Speech Science(DSS)의 음성평가 결과가 갖는 의미를 알아보고자 이 연구를 하였다. 성대결절환자 여자 성인 25명을 대상으로 DSS를 이용한 음성평가와 청각심리적 검사방법인 GRBAS와의 관계를 비교 분석하였다. 청각심리적 검사인 GRBAS의 0, 1, 2, 3의 각 Grade에 따라 DSS의 음성평가의 결과와 비교하였다. DSS의 음성평가 결과로서 Grade가 0, 1, 2인 경우 총 15례중 1례를 제외한 모든 경우에 있어서 hoarseness, harshness, breathiness항목에서 정상소견을 나타냈으며 Grade가 3인 경우에는 총 10례 중 6례에서 hoarseness, harshness, breathiness항목에서 정상소견을 나머지 4례에서는 hoarseness에서 slight한 정도를 보여주었다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11a
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• pp.707-710
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• 2002

본 논문은 오디오와 비디오 정보의 융합을 통한 멀티 모달 음성 인식 시스템을 제안한다. 음성 특징 정보와 영상 정보 특징의 융합을 통하여 잡음이 많은 환경에서 효율적으로 사람의 음성을 인식하는 시스템을 제안한다. 음성 특징 정보는 멜 필터 캡스트럼 계수(Mel Frequency Cepstrum Coefficients: MFCC)를 사용하며, 영상 특징 정보는 주성분 분석을 통해 얻어진 특징 벡터를 사용한다. 또한, 영상 정보 자체의 인식률 향상을 위해 피부 색깔 모델과 얼굴의 형태 정보를 이용하여 얼굴 영역을 찾은 후 강력한 입술 영역 추출 방법을 통해 입술 영역을 검출한다. 음성-영상 융합은 변형된 시간 지연 신경 회로망을 사용하여 초기 융합을 통해 이루어진다. 실험을 통해 음성과 영상의 정보 융합이 음성 정보만을 사용한 것 보다 대략 5%-20%의 성능 향상을 보여주고 있다.

• Journal of Sasang Constitutional Medicine
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• v.11 no.1
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• pp.137-157
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• 1999

The purpose of this study is to help classifying Sasang Constitution through correlation with sound characteristic. This study was done it under the suppose that Sasang Constitution has correlation with sound spectrogram. The following result were obtained about correlation between sound spectrogram and Sasang Constitution by comparison and analysis 1. Soeumin answered his voice low tone, smooth and quiet in the survey. Soyangin answered his voice high, clear, fast and speaking random. Taeumin answered his voice low, thick and muddy. 2. Taeyangin was significantly slow compared with the others in the time of reading composition. Taeyangin was significantly slow compared with the others in Formant frequency 1. Taeyangin was significantly discriminated from Soeumin in Formant frequency 5. Taeyangin was significantly low compared with the others in Bandwidth 2. Soeumln was significantly low compared with Taeyangin in Pitch Maximum and Pitch Maximum-Pitch Minimum. Taeyangin was significantly high compared with the others in Energy mean. 3. In list of specification, the discrimination rate was higher than that by lists of 13 in the results of Multi-dimensional 4-class minimum-distance. The discrimination rate of three disposition except Soyangin was higher than that of four disposition in the results of One way ANOVA and Analysis of dis crimination in SPSS/PC+. In CART, the estimate rate of Sasang Constitution discrimination was higher than any other method. It is considered that there is a correlation between sound spectrogram and Sasang constitution according to the results. And method of Sasang constitution classification through sound spectrogram analysis can be one method as assistant for the objectification of Sasang constitution classification.

• Journal of the Korea Convergence Society
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• v.8 no.5
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• pp.13-17
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• 2017

Existing Speech feature extracting method in speech Signal, there are incorrect recognition rates due to incorrect speech which is not clear threshold value. In this article, the modeling method for improving speech recognition performance that combines the feature extraction for speech and silence characteristics normalized to the non-speech. The proposed method is minimized the noise affect, and speech recognition model are convergence of speech signal feature extraction to each speech frame and the silence feature normalization. Also, this method create the original speech signal with energy spectrum similar to entropy, therefore speech noise effects are to receive less of the noise. the performance values are improved in signal to noise ration by the silence feature normalization. We fixed speech and non speech classification standard value in cepstrum For th Performance analysis of the method presented in this paper is showed by comparing the results with CHMM HMM, the recognition rate was improved 2.7%p in the speech dependent and advanced 0.7%p in the speech independent.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.3
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• pp.61-67
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• 2001

In this paper, a method that apply LP algorithm to image for speech recognition is suggested, using both speech and image information for recogniton of korean numeral speech. The input speech signal is pre-emphasized with parameter value 0.95, analyzed for B th LP coefficients using Hamming window, autocorrelation and Levinson-Durbin algorithm. Also, a gray image signal is analyzed for 2-dimensional LP coefficients using autocorrelation and Levinson-Durbin algorithm like speech. These parameters are used for input parameters of neural network using back-propagation algorithm. The recognition experiment was carried out at each noise level, three numeral speechs, '3','5', and '9' were enhanced. Thus, in case of recognizing speech with 2-dimensional LP parameters, it results in a high recognition rate, a low parameter size, and a simple algorithm with no additional feature extraction algorithm.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.462-472
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• 2004

In this paper a method for compensating for thp performance degradation or automatic speech recognition (ASR) is proposed. which is mainly caused by speaking rate variation. Before the new method is proposed. quantitative analysis of the performance of an HMM-based ASR system according to speaking rate is first performed. From this analysis, significant performance degradation was often observed in the rapidly speaking speech signals. A quantitative measure is then introduced, which is able to represent speaking rate. Time scale modification (TSM) is employed to compensate the speaking rate difference between input speech signals and training speech signals. Finally, a method for compensating the performance degradation caused by speaking rate variation is proposed, in which TSM is selectively employed according to speaking rate. By the results from the ASR experiments devised for the 10-digits mobile phone number, it is confirmed that the error rate was reduced by 15.5% when the proposed method is applied to the high speaking rate speech signals.

• The Journal of the Acoustical Society of Korea
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• v.41 no.3
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• pp.367-374
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• 2022

In this paper, correlation analysis between speech characteristics and the dose of antipsychotic drugs was performed. To investigate the pattern of speech characteristics of ExtraPyramidal Symptoms (EPS) related to voice change, a common side effect of antipsychotic drugs, a Korean-based extrapyramidal symptom speech corpus was constructed through the sentence development. Through this, speech patterns of EPS and non-EPS groups were investigated, and in particular, a strong speech feature correlation was shown in the EPS group. In addition, it was confirmed that the type of speech sentence affects the speech feature pattern, and these results suggest the possibility of early detection of antipsychotics-induced EPS based on the speech features.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.12
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• pp.1187-1192
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• 2016

Performance analysis results are limited to of network level, because network level transmission parameters are used for performance measure and analysis of network design, construction and operation on underwater MANET, With this way of performance analysis based on network level, it is not easy to analyze transmission performance related with user level transmission quality. In this paper, transmission performance focused on application traffic be required by user is investigated to supplement weakness of performance analysis based on network level. Voice traffic, which is expected to be increasingly used on underwater MANET, is considered as application service, Some conditions for underwater MANET will be proposed to support transmission quality, MOS, CCR and EED, etc.. A computer simulation based on NS-2 is used for performance measure, voice traffic is generated as VoIP specification.