• Journal of the Korea Society of Computer and Information
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• v.5 no.3
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• pp.91-98
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• 2000

In this paper we design implement the speaker verification system for verifying personal identification using voice. Filter bank magnitude was used as a feature parameter and code-book was made using LBG a1gorithm. The code book convert feature parameters into code sequence. The difference between reference pattern and input pattern measures using DTW(Dynamic Time Warping). The similarity measured using DTW and threshold value derived from deviation were used to discriminate impostor from client speaker.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.107-110
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• 1998

본 논문은 DTW(Dynamic Time Warping)을 이용한 음성인식에서 표준패턴(reference patterns)으로 사용되는 벡터열을 GA(Genetic Algorithm)을 이용하여 보다 적응된 패턴의 벡터열로 생성하는 방법을 제시한다. 본 논문의 필요성은 다음과 같다. 음성인식의 주요한 엔진들 중에 하나로 DTW가 사용된다[1]. DTW는 표준패턴과 시험패턴(test patterns)간의 최적 경로(optimal path)를 찾아내어 가장 유사한 패턴을 찾아내는 방법을 말한다. 그러나 음성은 같은 발음에 대해서도 사람의 발성 길이와 목의 상태 등에 따라 다양한 패턴으로 나타나며 동일 화자의 같은 어휘도 시간과 환경에 따라 변한다. 따라서 이러한 음성의 동적 특성에 적응하는 방법이 필요하다. 본 논문은 이러한 문제에 대한 해결 방법으로 GA를 이용하여 보다 적합하고 적응적인 표준 패턴을 생성시켜 적응하는 방법을 개발하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.123-126
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• 1998

기존의 DTW를 이용한 화자 인식 시스템은 DTW의 단점이라 할 수 있는 과다한 계산량을 갖는다는 문제점을 갖고 있다. 따라서 본 논문은 텍스트 종속 화자 인식 시스템에서 피치 분포도를 갖는 개별 화자의 lDTW를 수행하기 전에 시간축 스케일링을 이용한 전처리로 인식시의 계산량을 감소시키는 과정을 미리 수행할 후 감소된 기준패턴들의 입력신호에 대해서만 DTW를 수행하는 방법을 제안하고자 한다. 제안한 방법을 실험하였을 경우 87.5%의 평균 처리 시간이 감소하였고, 더불어 인식률 감소는 거의 없었다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.430-437
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• 2002

Having a variety of good characteristics against other pattern recognition techniques, the multilayer perceptron (MLP) has been widely used in speech recognition and speaker recognition. But, it is known that the error backpropagation (EBP) algorithm that MLP uses in learning has the defect that requires restricts long learning time, and it restricts severely the applications like speaker recognition and speaker adaptation requiring real time processing. Because the learning data for pattern recognition contain high redundancy, in order to increase the learning speed it is very effective to use the online-based learning methods, which update the weight vector of the MLP by the pattern. A typical online EBP algorithm applies the fixed learning rate for each update of the weight vector. Though a large amount of speedup with the online EBP can be obtained by choosing the appropriate fixed rate, firing the rate leads to the problem that the algorithm cannot respond effectively to different learning phases as the phases change and the number of patterns contributing to learning decreases. To solve this problem, this paper proposes a Changing rate and Omitting patterns in Instant Learning (COIL) method to apply the variable rate and the only patterns necessary to the learning phase when the phases come to change. In this paper, experimentations are conducted for speaker verification and speech recognition, and results are presented to verify the performance of the COIL.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.36T no.4
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• pp.30-37
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• 1999

This study suggested the Fuzzy Lyapunov dimension. The Fuzzy Lyapunov dimension is to evaluate the quantitative variation of the attractor. In this paper the speaker recognition is evaluated by the Fuzzy Lyapunov dimension. It has been proved that the suggested Fuzzy Lyapunov dimension is superior in the discrimination characteristics between standard reference pattern attractors, and in reference to the test pattern attractor, it has been verified that it is the speaker recognition parameter which absorbs the pattern variation. In order to evaluate the Fuzzy Lyapunov dimension as speaker recognition parameter, the mistaken recognition according to discrimination error in each of speaker and standard reference pattern was estimated, and the validity of the speaker recognition parameter was experimental. As the result of the speaker recognition experiment, 97.0[%] of recognition ratio was obtained, and it was confirmed that the Fuzzy Lyapunov dimension was fit for the speaker recognition parameter.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.146-156
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• 1989

This paper discusses how to choose speech parameters and decision logics to improve the performance of speaker recognition systems. It also considers the influence of the reference patterns on the speaker recognition. It is observed from the performance analysis based on LPSs, PARCOR coefficients and LPC-cepstrum coefficients that LPC-cepstrum coefficients are superior to the others in speaker recognition without regard to the reference patterns. In order to improve the recognition performance, a new decision logic is proposed based on a generalized-distance concept. It differs from the existing methods in that it considers the statistics of customer and impostors at the same time. It turns out from a speaker verification test that the proposed decision logic ferforms better than the existing ones.

• Journal of the Korea Society of Computer and Information
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• v.8 no.3
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• pp.73-79
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• 2003

This paper proposes speaker recognition algorithm which includes both the pitch detection and the fuzzified pattern matching. This study utilizes pitch pattern using a pitch and speech parameter uses binary spectrum. In this paper. makes reference pattern using fuzzy membership function in order to include time variation width for non-utterance time and performs vocal track recognition of common character using fuzzified pattern matching.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.131-134
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• 1998

본 논문에서는 전자 개인 비서 시스템(PEAS)의 일부인 일정관리 시스템을 화자인식 기술을 적용하여 구현하였다. 본 시스템은 음성을 패스워드로 개인을 확인하여 각 개인의 일정을 관리해 주는 것으로 보안성과 함께 사용자에게 편의성을 제공한다. 사용자 등록을 자유롭게 하였으며 인식에서는 계산 시간 등을 고려하여 DTW 알고리즘에서 얻을 수 있는 경로정보를 이용해 하나의 참조패턴을 구성하도록 하였다. 또한 시간 흐름에 따라 인식율 저하를 방지하기 위해 실험결과에 따라 일정기간 뒤에 자동으로 참조패턴이 갱신되도록 하였다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2003.05a
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• pp.153-156
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• 2003

본 연구는 화자종속 소어휘 음성인식의 성능을 개선하는 데 그 목적이 있다. 인식에 사용될 음성의 특징을 얻기 위해 Winer 필터와 LPC＆Cepstrum을 이용하여 프레임 당 12차 패턴을 추출하였다. 추출된 특징패턴을 인식하는 인식부는 특히 소어휘 음성인식에 우수한 성능을 보이는 기존의 역전파 신경회로망(Backpropagation Neural Network)에 인식율 개선을 위하여 퍼지추론시스템을 결합한 형태로 구현되었다. 실험결과 신경망만을 사용한 경우에 비하여 인식율이 향상됨을 연구하였다.

• Journal of the Korea Computer Industry Society
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• v.2 no.4
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• pp.471-480
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• 2001

In this thesis, we perform the experiment of speaker recognition by identifying vowels in the pronunciation of each speaker using Adaptive Resource Theory 2(ART2) model. The 5 adult males and 5 adult females pronounce from 0 to 9 digits. We extract the vowels from the pronunciation of each speaker first, we are extracted characteristic coefficient through a pitch detection algorithm, a LPC analysis, and a LPC cepstral analysis to generate an input pattern of ART2. The experimental results showed that pitch coefficients are somewhat more enhanced than LPC or LPC cepstral coefficient.