• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.5
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• pp.472-481
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• 1992

In this paper, we proposed a method of segmenting speech signal by neural network and its validity is proved by computer simulation. The neural network Is composed of multi layer perceptrons with one hidden layer. The matching accuracies of the proposed algorithm are measured for continuous vowel and place names. The resulting average matching accuracy is 100% for speaker-dependent case, 99.5% for speaker-independent case and 94.5% for each place name when the neural network 1,; trained for 6 place names simultaneously.

• Journal of Digital Convergence
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• v.13 no.10
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• pp.353-358
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• 2015

The Vocabulary Recognition System made by recognizing the standard vocabulary is seen as a decline of recognition when out of the standard or similar words. In this case, reconstructing the system in order to add or extend a range of vocabulary is a way to solve the problem. This paper propose configured Bhattacharyya algorithm standing by speech recognition learning model using the Bayesian methods which reflect parameter estimation upon the model configuration scalability. It is recognized corrected standard model based on a characteristic of the phoneme using the Bayesian methods for parameter estimation of the phoneme's data and Bhattacharyya algorithm for a similar model. By Bhattacharyya algorithm to configure recognition model evaluates a recognition performance. The result of applying the proposed method is showed a recognition rate of 97.3% and a learning curve of 1.2 seconds.

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.241-244
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• 1994

이 논문은 대어휘, 화자독립 음성인식 시스템인 KT-STOCK과 이 시스템에 대한 전화망을 통한 실험적 실용시험에 대해 기술하였다. KT-STOCK은 현재 주식시장에 상장된 712개 회사의 현재주가를 음성을 이용하여 검색할 수 있는 시스템이다. 이 시스템은 hidden markov model 기술에 기반을 둔 고립단어 인식 시스템이며 유사음소를 기본 인식단위로 사용한다. KT-STOCK은 1994년 6월 24일부터 실험적 실용시험 중에 있다. 중간 결과에 따르면 모의 실험 결과는 실제 환경에서의 시험과 차이가 있는 거승로 나타났다. 실제 환경에서 이 시스템의 인식률은 현재 61.9%이다.

• ETRI Journal
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• v.8 no.1
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• pp.6-15
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• 1986

본고는 보상훈련 기간 중 일본 동북대학 응용정보학 연구센타에서 수행한 연구 결과를 기술한 것이다. 음소 단위에 의한 한국어의 대용량 단어인식을 위한 기초연구로서, 그 기본이 되는 단모음을 대상으로 포먼트 주파수에 의한 음운간의 특징 및 발성자간의 개인성의 분산을 살펴보고 Battacharyya 거리를 구하여 음운간의 식별의 곤란성을 도출하였다. 또한, Karbunen-Loeve변환 및 Bayes결정에 의한 인식 그리고 spectral local peak에 의한 인식등의 실험에 의해 효과적인 인식 방법에 관하여 고찰하였다 .

• Electronics and Telecommunications Trends
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• v.5 no.2
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• pp.63-77
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• 1990

HMM(Hidden Markov Model)은 음성을 기술하는데 적합한 model이다. 본 고는 최근 CMU에서 개발한 HMM에 기반을 둔 화자독립, 연속음성 system인 SPIHNX에 대하여 기술한다. SPHINX는 단순한 음소의 HMM model을 적용한 baseline SPHINX로부터 시작하여 새로운 지식의 추가 및 음성단위의 조정 등을 통하여 지속적으로 그 성능이 개선되어 왔다. SPHINX의 최종 version은 어휘 약 1000단어 정도의 재원 관리에 관한 질문 형태의 문장을 인식하는데 96%의 높은 인식율을 보인다. SPHINX는 가장 발전된 음성인식 시스템의 하나이며 이는 화자독립, 대용량어휘의 연속음성 인식 시스템의 실현 가능성을 제시한다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.27-34
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• 2001

Utterance verification is used in variable vocabulary word recognition to reject the word that does not belong to in-vocabulary word or does not belong to correctly recognized word. Utterance verification is an important technology to design a user-friendly speech recognition system. We propose a new utterance verification algorithm for no-training utterance verification system based on the minimum verification error. First, using PBW (Phonetically Balanced Words) DB (445 words), we create no-training anti-phoneme models which include many PLUs(Phoneme Like Units), so anti-phoneme models have the minimum verification error. Then, for OOV (Out-Of-Vocabulary) rejection, the phoneme-based confidence measure which uses the likelihood between phoneme model (null hypothesis) and anti-phoneme model (alternative hypothesis) is normalized by null hypothesis, so the phoneme-based confidence measure tends to be more robust to OOV rejection. And, the word-based confidence measure which uses the phoneme-based confidence measure has been shown to provide improved detection of near-misses in speech recognition as well as better discrimination between in-vocabularys and OOVs. Using our proposed anti-model and confidence measure, we achieve significant performance improvement; CA (Correctly Accept for In-Vocabulary) is about 89％, and CR (Correctly Reject for OOV) is about 90％, improving about 15-21％ in ERR (Error Reduction Rate).

• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.3-10
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• 1999

This paper describes the development and evaluation of a Korean address input system employing automatic speech recognition technique as user interface for input Korean address. Address consists of cities, provinces and counties. The system works on a window 95 environment of personal computer with built-in soundcard. In the speech recognition part, the Continuous density Hidden Markov Model(CHMM) for making phoneme like units(PLUs) and One Pass Dynamic Programming(OPDP) algorithm is used for recognition. For address recognition, Finite State Automata(FSA) suitable for Korean address structure is constructed. To achieve an acceptable performance against the variation of speakers, microphones, and environmental noises, Maximum a posteriori(MAP) estimation is implemented in adaptation. And to improve the recognition speed, fast search method using variable pruning threshold is newly proposed. In the evaluation tests conducted for the 100 connected words uttered by 3 males the system showed above average 96.0% of recognition accuracy for connected words after adaption and recognition speed within 2 seconds, showing the effectiveness of the system.

• The Journal of the Acoustical Society of Korea
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• v.12 no.4
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• pp.57-62
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• 1993

본 논문은 신경망을 이용한 우리말 모음에 대한 인식 연구이다. 음성을 나누거나. 음소별 인식이나, 시간 신축 방법을 사용하지 않고 모음을 인식하였다. 식나의 변화에 따른 음성의 변화를 정적인 음성으로 취급하였다. 10개로 균등히 나눈 프레임에 각 프레임마다 10차의 PARCOR계수를 추출하였다. 신경망의 구조를 간단히 하기 위해서 단모음과 복모음을 구분하여 학습시켰으며, 출력 노드의 수를 감소시키기 위해 이진 코드 형태로 구성하였다.

• Information and Communications Magazine
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• v.7 no.3
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• pp.16-26
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• 1990

본 논문은 음성인식을 위한 neural network의 하나인 TRACE model에 대하여 기술한다. TRACE model은 HEARSAY 음성이해 system의 blackboard 구조와 유사한 trace 구조를 사용하여 정보처리 unit들간의 자극성 내지 억제성 연결에 의한 상호작용의 결과로서 음성을 인식한다. TRACE model은 특징층, 음소층 그리고 단어층의 3층으로 구성되며 층간의 unit들이 상호작용할 뿐 아니라 동일층의 unit들이 상호경쟁함으로써 음성인식에 있어서의 context 효과, segmentation 및 잡음 등의 문제를 해결할 수 있다.