• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.60-65
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• 2002

In this paper, we propose GFCC(gammatone filter frequency cepstrum coefficient) parameter which was based on the auditory characteristic for accomplishing better speech recognition rate. And it is performed the experiment of speech recognition for isolated word acquired from telephone network. For the purpose of comparing GFCC parameter with other parameter, the experiment of speech recognition are carried out using MFCC and LPCC parameter. Also, for each parameter, we are implemented CMS(cepstral mean subtraction)which was applied or not in order to compensate channel distortion in telephone network. Accordingly, we found that the recognition rate using GFCC parameter is better than other parameter in the experimental result.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.935-938
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• 2000

본 논문에서는 음성 인식을 이용하여 전화를 교환해주는 자동 교환 시스템을 구현하고, 성능을 평가하였다. 구현된 시스템에는 필수적인 음성인식 이외에도 DSP 진단 기능, 인식 대상 어휘의 추가 및 변경기능, 음성 수집 기능 등이 구현 되었다. SCHMM (Semi-Continuous Hidden Markov Model)을 이용한 전화망에서의 화자 독립 고립 단어 가변 어휘 인식을 대상으로 하였으며, 실시간 구현을 위하여 Texas Instrument 사의 TMS320C32를 사용하였다〔6〕. 인식 어휘는 부서명 및 인명이고 1300여 단어일 때, 인식 성능은 91.5%이다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2000.11a
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• pp.209-212
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• 2000

본 논문에서 마우스 대신에 음성으로 명령을 입력하여 퍼지 추론을 통해 윈도우 화면상의 커서를 이동시키는 인터페이스를 구현하였다. 입력된 음성이 대체로 짧은 언어이기에 이를 인식하기 위하여 고립단어 인식에 강한 DTW방식을 사용하였다. DTW방식의 단점중인 하나가 음성길이가 비슷한 명령을 입력하였을 때 표준패턴 중 오차 값이 가장 작은 패턴으로 인식하는 것이다. 예를들면 아주 많이 이동해 라는 음성이 입력되었을 때 동일한 음성길이를 가진 아주 많이 오른쪽으로 인식하는 경우가 있다. 이런 오류를 해결하고자 각 패턴의 DTW 오차 값 범위와 표준 패턴의 음성길이를 기준으로 임계값을 퍼지 추론하여 명령으로서 수락 여부를 결정하였다. 판단이 애매한 부분은 사용자에게 질의를 하여 응답에 따라 수락 여부를 결정하였다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.6
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• pp.64-70
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• 1991

본논문에서는 Hidden Markov Model 에 기초한 실시간 고립 단어 인식 시스템에서의 Viterbi 알 고리듬을 위한 전용 VLSI 구조를 제안하였다. 제안된 구조는 듀얼포트 레지스터 파일로 입출력 부하를 줄이고 가산-최소/최대 연산부의 병렬 연산 구조를 이용하여 실시간 동작이 가능하도록 설계되었다. 모 델 인자와 상태 변수의 값에 태그들을 덧붙임으로써 이 구조는 대표적인 HMM 구조들을 쉽게 구현할 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.9
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• pp.738-744
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• 1990

This paper proposed a real-time memory storage method and an improved sample data method from given data of the speech signal, so, the isolated word recognition system using a back-propagation learning algorithm of the neural netwrok is studied. The recognition rate and the error rate are compared with the new sample data sets generated from small sets of given sample data by the node nunber variatiion of each layer. In this result, the recognition rate of 95.1% was achived.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2001.12a
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• pp.157-160
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• 2001

본 논문에서는 강수량과 자동차 주행속도 등의 환경조건에 따라 와이퍼 속도를 일정하게 적용한 기존의 시스템을 개선하여 운전자의 개인 특성에 의해서도 속도 변경이 가능하게 함으로서 인간에게 조금 더 친밀감을 제공하는 시스템을 구현하였다. 초기 와이퍼 속도는 입력받은 강수량과 자동차 주행 속도로 추론하여 구하였다. 추론된 와이퍼 속도를 운전자의 개인 특성에 따라 변경하고자 할 경우, 해당 음성명령을 입력받아 재 추론하였다. 음성인식을 위해서는 고립단어 인식에 적절한 DTW방식을 사용하였고, 와이퍼 속도는 퍼지 추론을 적용하여 구하였다.

• The Journal of the Korea Contents Association
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• v.8 no.1
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• pp.328-338
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• 2008

Most speech recognition systems that have a large capacity and high recognition rates are isolated word speech recognition systems. In order to extend the scope of recognition, it is necessary to increase the number of words that are to be searched. However, it shows a problem that exhibits a decrease in the system performance according to the increase in the number of words. This paper defines the context information that affects speech recognition in a ubiquitous environment to solve such a problem and develops user localization method using inertial sensor and RFID. Also, we develop a new speech recognition system that demonstrates better performances than the existing system by establishing a word model domain of a speech recognition system by context information. This system shows operation without decrease of recognition rate in smart home environment.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.74-81
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• 1995

This paper describes the development of speech databases for the Korean language which were constructed at Communications Research Laboratory in KAIST. The procedure and environment to construct the speech database are presented in detail, and the phonetic and linguistic properties of the databases are presented. the databases were intended for use in designing and evaluating speech recognition algorithms. The databases consist of five different sets of speech contents : trade-related continuous speech with 3,000 words, variable-length connected digits, phoneme-balanced 75 isolated words, 500 isolated Korean provincial names, and Korean A-set words.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.5-10
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• 1995

In this paper we present a clustering procedure based on the use of HMM in order to get multiple statistical models which can well absorb the variants of each speaker with different ways of saying words. The HMM-clustered models obtained from the developed technique are applied to the speaker independent isolated word recognition. The HMM clustering method splits off all observation sequences with poor likelihood scores which fall below threshold from the training set and create a new model out of the observation sequences in the new cluster. Clustering is iterated by classifying each observation sequence as belonging to the cluster whose model has the maximum likelihood score. If any clutter has changed from the previous iteration the model in that cluster is reestimated by using the Baum-Welch reestimation procedure. Therefore, this method is more efficient than the conventional template-based clustering technique due to the integration capability of the clustering procedure and the parameter estimation. Experimental data show that the HMM-based clustering procedure leads to $1.43\%$ performance improvements over the conventional template-based clustering method and $2.08\%$ improvements over the single HMM method for the case of recognition of the isolated korean digits.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.567-575
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• 2001

This paper suggests new method to get fixed size parameter from different length of voice signals. The efficiency of speech recognizer is determined by how to compare the similarity(distance of each pattern) of the parameter from voice signal. But the variation of voice signal and the difference of speech speed make it difficult to extract the fixed size parameter from the voice signal. The method suggested in this paper is to normalize the parameter at fixed size by using the 2 dimension DCT(Discrete Cosine Transform) after representing the parameter by spectrogram. To prove validity of the suggested method, parameter extracted from 32 auditory filter-bank(it estimates auditory nerve firing probabilities) is used for the input of neural network after being processed by 2 dimension DCT. And to compare with conventional methods, we used one of conventional methods which solve time alignment problem. The result shows more efficient performance and faster recognition speed in the speaker dependent and independent isolated word recognition than conventional method.