• Journal of Digital Convergence
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• v.10 no.10
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• pp.529-538
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• 2012

This study analyzed a variety of different kinds of usage patterns of four-digit personal identification numbers(PINs) based on the data collected from students attending at A-university in 2011. According to the analysis, the 4 digit PINs '1234' was being used most frequently which is similar to the findings of the study by the Daniel Amitay research. In addition, almost 50 percent of the students were using the four-digit PINs closely related to their personal information, and more than 10 percent of them were using it only by the reason of easiness to remember or convenience to use. Number '0' was the most frequently used, and girl students used it more than boys did. According to the result of analysis of four-digit PINs, it was found that number '1' was the most frequently used in the first and the third position. It was also found that number '2' and number '4' were the most frequently used in the second and the forth position respectively. Among students who had used number 3 in third position, around 63.6 percent of those students were using number 4 in the forth position in their PINs which showed highest frequency from analysis of double-digits connected together in each position. The analysis of PINs were represented by the polygonal (type) graph with that X-axis showed from first to forth position and Y-axis showed number had been used at each position. Among many polygonal graphs, the form showed an N-type which was called in type 1 took the first place by 35 percent.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.44-47
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• 1999

본 논문에서는 한국어 연결숫자인식에 있어서 모델간의 변별력 향상을 통해 인식률을 높이기 위하여 음향학적 파라메터(Acousticparameter)를 사용하는 짓을 제안한다. 제안된 방법은 음성학적 지식에 근거하여 적절한 주파수 대역별 에너지의 비의 로그값을 추가적인 특징파라메터로 사용한다. 실험결과, 제안된 방법을 사용함으로써 기본 인식시스템에 비해 오류율이 최고 $46\%$ 정도 감소됨을 확인할 수 있었다. 그리고 채널보상 기술을 함께 적용함으로써 $69\%$ 정도의 오류율 감소를 얻었다.

• Proceedings of the KIEE Conference
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• 1987.11a
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• pp.338-340
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• 1987

The speech recognition system usually consists of two modules, segmentation module and identification module. So, the performance of the system heavily depends on the segmentation accuracy and the segmentation unit. This paper is concerned with the agreeable features for segmentation in syllables. Total energy and two band width energy. (LE:4000-5000Hz and HE:900-3100Hz) are suitable cues for segmentation. And we testify it through the experiment using connected digit.

• Proceedings of the KSPS conference
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• 2003.05a
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• pp.145-148
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• 2003

In this paper, we carried out comparative study about various feature parameters for the effective speaker recognition such as LPC, LPCC, MFCC, Log Area Ratio, Reflection Coefficients, Inverse Sine, and Delta Parameter. We also adopted cepstral liftering and cepstral mean subtraction methods to check their usefulness. Our recognition system is HMM based one with 4 connected-Korean-digit speech database. Various experimental results will help to select the most effective parameter for speaker recognition.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11a
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• pp.212-215
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• 2003

본 논문에서는 숫자음과 단위음으로 구성된 한국어 연결 단위숫자음 인식의 성능 향상을 위하여 한국어 연결 단위숫자음의 특징을 분석하였다. 한국어의 단위숫자음은 숫자음 한음절과 단위음 한음절로 구성된 두음절의 연속적이고 반복적인 발성으로 나타난다. 숫자음에서의 인식 대상 어휘는 숫자음이라는 제한된 규칙을 갖는 가변 숫자음이다. 따라서 개수, 금액, 단위량, 거래량 등에서 나타날 수 있는 가변 숫자음을 인식하기 위하여 FSN(Finite State Network)을 구성하였다. 음향 모델은 한국어 숫자음과 같이 발성구간이 짧은 어휘의 연결음 (connected word)의 인식에서 효과적인 반음절쌍(demi-syllable pair) 모델을 이용하였다 실험결과, 화자 독립적인 가변 숫자음 60문장의 테스트 데이터에 대해서 문장 인식률 91.0%로 인식 성능을 향상시킬 수 있었다.

• Proceedings of the KIEE Conference
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• 1989.07a
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• pp.481-485
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• 1989

The object of this parer is on the design of an broad board classifier for connected. Korean digit. Many approaches have been applied in speech recognition systems: parametric vector quantization, dynamic programming and hiden Markov model. In the 80's the neural network method, which is expected to solve complex speech recognition problems, came bach. We have chosen the rule based system for our model. The phoneme-groups that we wish to classify are vowel_like, plosive_like fricative_like, and stop_like.The data used are 1380 connected digits spoken by three untrained male speakers. We have seen 91.5% classification rate.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.8
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• pp.96-105
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• 1998

The aim of this paper is to propose the method of WDHMM(Weighted DHMM), using the MLP-VQ for the improvement of speaker-independent connect-digit recognition system. MLP neural-network output distribution shows a probability distribution that presents the degree of similarity between each pattern by the non-linear mapping among the input patterns and learning patterns. MLP-VQ is proposed in this paper. It generates codewords by using the output node index which can reach the highest level within MLP neural-network output distribution. Different from the old VQ, the true characteristics of this new MLP-VQ lie in that the degree of similarity between present input patterns and each learned class pattern could be reflected for the recognition model. WDHMM is also proposed. It can use the MLP neural-network output distribution as the way of weighing the symbol generation probability of DHMMs. This newly-suggested method could shorten the time of HMM parameter estimation and recognition. The reason is that it is not necessary to regard symbol generation probability as multi-dimensional normal distribution, as opposed to the old SCHMM. This could also improve the recognition ability by 14.7% higher than DHMM, owing to the increase of small caculation amount. Because it can reflect phone class relations to the recognition model. The result of my research shows that speaker-independent connected-digit recognition, using MLP-VQ and WDHMM, is 84.22%.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.764-767
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• 2004

The performance of a speech recognition system is generally degraded in telephone environment because of distortions caused by background noise and various channel characteristics. In this paper, data-driven temporal filters are investigated to improve the performance of a specific recognition task such as telephone speech. Three different temporal filtering methods are presented with recognition results for Korean connected-digit telephone speech. Filter coefficients are derived from the cepstral domain feature vectors using the principal component analysis.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.557-560
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• 2003

Voice is one of the promising biometrics because it is one of the most convenient ways human would distinguish someone from others. The target of speaker verification is to divide the client from imposters. Support Vector Machine(SVM) is in the limelight as a binary classifier, so it can work well in speaker verification. In this paper, we combined SVM with genetic algorithm(GA) to reduce the dimensionality of input feature. Experiments were conducted with Korean connected digit database using different feature dimensions. The verification accuracy of SVM with GA is slightly lower than that of SVM, but the proposed algorithm has greater strength in the memory limited systems.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.250-253
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• 1998

In digital communicatioin networks, speech recognition systems conventionally reconstruct speech followed by extracting feature [parameters. In this paper, we consider a useful approach by incorporating speech coding parameters into the speech recognizer. Most speech coders employed in the networks represent line spectral pairs as spectral parameters. In order to improve the recognition performance of the LSP-based speech recognizer, we introduce two different ways: one is to devise weighed distance measures of LSPs and the other is to transform LSPs into a new feature set, named a pseudo-cepstrum. Experiments on speaker-independent connected-digit recognition showed that the weighted distance measures significantly improved the recognition accuracy than the unweighted one of LSPs. Especially we could obtain more improved performance by using PCEP. Compared to the conventional methods employing mel-frequency cepstral coefficients, the proposed methods achieved higher performance in recognition accuracies.