• 대한음성학회지:말소리
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• 제53호
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• pp.143-156
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• 2005

Speaker adaptation in eigenvoice space is a popular method for rapid speaker adaptation. To improve the performance of the method, the number of speaker dependent models should be increased and eigenvoices should be re-estimated. However, principal component analysis takes much time to find eigenvoices, especially in a continuous speech recognition system. This paper describes a method to reduce computation time to estimate eigenvoices only for supplementary speaker dependent models and to merge them with the used eigenvoices. Experiment results show that the computation time is reduced by 73.7% while the performance is almost the same in case that the number of speaker dependent models is the same as used ones.

• 한국음향학회지
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• 제15권4호
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• pp.5-11
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• 1996

본 연구에서는 음절 단위의 HMM을 이용하여 발성한 문장에 대해 화자 적응화 할 수 있는 방법을 제안하였다. 문장에 대한 음절 단위의 추출은 음절HMM의 연결과 viterbi세그멘테이션으로 자동화하였고, 화자 적응화는 소량의 문장과 문장의 추가에서도 시켄셜적으로 적응화할 수 있는 MAPE(최대 사후 확률 추정)를 이용한 학습으로 수행하였다. 신문 사설에서 취한 문장에 대하여 화자 적응화한 경우의 인식을 71.8%로 적응화 전의 결과보다 37% 향상되었다.

• 대한음성학회지:말소리
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• 제60호
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• pp.109-123
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• 2006

Since computing the log-likelihood of Gaussian mixture density is a major computational burden for the speech recognizer based on the continuous HMM, several techniques have been proposed to reduce the number of mixtures to be used for recognition. In this paper, we propose a modified Bucket Box Intersection (BBI) algorithm, in which two relative thresholds are employed: one is the relative threshold in the conventional BBI algorithm and the other is used to reduce the number of the Gaussian boxes which are intersected by the hyperplanes at the boxes' edges. The experimental results show that the proposed algorithm reduces the number of Gaussian mixtures by 12.92% during the recognition phase with negligible performance degradation compared to the conventional BBI algorithm.

• 한국산업정보학회논문지
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• 제7권3호
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• pp.74-80
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• 2002

한국어 연속 음성 인식결과의 성능향상을 위해서 자연어 처리 기술을 이용한 후처리 기법이 사용된다. 그러나 자연어 처리 기법이 대부분 띄어쓰기가 있는 정형화된 입력 문장에 대한 분석을 수행하여 왔기 때문에 형태소 분석기를 직접 음성인식 결과의 향상에 사용하는 데에는 어려운 점이 많다. 본 논문에서는 띄어쓰기를 고려하지 않는 기능어 기반의 최장일치 형태소 해석 방법인 구문 형태소 단위의 분석을 이용한 음정인식 결과의 향상 모델을 제안한다. 제안된 모델을 통해 연속음성 인식 결과에서 자주 발생하는 용언과 보조 용언 및 의존 명사 사이의 음운들 사이의 구조적 정보를 활용함으로써 음성 인식 결과의 성능을 향상시키는 방법에 대해 기술한다.

• The Journal of the Acoustical Society of Korea
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• 제17권4E호
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• pp.11-16
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• 1998

For large vocabulary speech recognition using HMMs, context-dependent subword units have been often employed. However, when context-dependent phone models are used, they result in a system which has too may parameters to train. The problem of too many parameters and too little training data is absolutely crucial in the design of a statistical speech recognizer. Furthermore, when building large vocabulary speech recognition systems, unseen triphone problem is unavoidable. In this paper, we propose the modified phonetic decision tree algorithm for the automatic prediction of unseen triphones which has advantages solving these problems through following two experiments in Japanese contexts. The baseline experimental results show that the modified tree based clustering algorithm is effective for clustering and reducing the number of states without any degradation in performance. The task experimental results show that our proposed algorithm also has the advantage of providing a automatic prediction of unseen triphones.

• 음성과학
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• 제10권1호
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• pp.71-84
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• 2003

In the acoustic modeling for large vocabulary speech recognition, a sparse data problem caused by a huge number of context-dependent (CD) models usually leads the estimated models to being unreliable. In this paper, we develop a new clustering method based on the C45 decision-tree learning algorithm that effectively encapsulates the CD modeling. The proposed scheme essentially constructs a supervised decision rule and applies over the pre-clustered triphones using the C45 algorithm, which is known to effectively search through the attributes of the training instances and extract the attribute that best separates the given examples. In particular, the data driven method is used as a clustering algorithm while its result is used as the learning target of the C45 algorithm. This scheme has been shown to be effective particularly over the database of low unknown-context ratio in terms of recognition performance. For speaker-independent, task-independent continuous speech recognition task, the proposed method reduced the percent accuracy WER by 3.93% compared to the existing rule-based methods.

• 음성과학
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• 제15권2호
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• pp.67-77
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• 2008

In this paper, we implemented a robust speech recognizer using the TMS320VC33 DSP. For this implementation, we had built speech and noise database suitable for the recognizer using spectral subtraction method for noise removal. The recognizer has an explicit structure in aspect that a speech signal is enhanced through spectral subtraction before endpoints detection and feature extraction. This helps make the operation of the recognizer clear and build HMM models which give minimum model-mismatch. Since the recognizer was developed for the purpose of controlling car facilities and voice dialing, it has two recognition engines, speaker independent one for controlling car facilities and speaker dependent one for voice dialing. We adopted a conventional DTW algorithm for the latter and a continuous HMM for the former. Though various off-line recognition test, we made a selection of optimal conditions of several recognition parameters for a resource-limited embedded recognizer, which led to HMM models of the three mixtures per state. The car noise added speech database is enhanced using spectral subtraction before HMM parameter estimation for reducing model-mismatch caused by nonlinear distortion from spectral subtraction. The hardware module developed includes a microcontroller for host interface which processes the protocol between the DSP and a host.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.61-65
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• 1992

This paper presents a study on Neural Networks for Phoneme Recognition and performs phoneme recognition using TDNN(Time Delay Neural Network). Also, this paper proposes new training algorithm for speech recognition using neural nets that proper to large scale TDNN. Because phoneme recognition is indispensable for continuous speech recognition, this paper uses TDNN to get accurate recognition result of phoneme. And this paper proposes new training algorithm that can converge TDNN to optimal state regardless of the number of phoneme to be recognized. The result of recognition on three phoneme classes shows recognition rate of 9.1%. And this paper proves that proposed algorithm is a efficient method for high performance and reducing convergence time.

• 전자공학회논문지B
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• 제32B권1호
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• pp.171-179
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• 1995

In many continuous speech recognition systems, phoneme is used as a basic recognition unit However, the coarticulation generated among neighboring phonemes makes difficult to recognize phonemes consistently. This paper proposes allophone as an alternative recognition unit. We have classified each phoneme into three different allophone groups by the location of phoneme within a syllable. For a recognition algorithm, time-delay neural network(TDNN) has been designed. To recognize all Korean allophones, TDNNs are constructed in modular fashion according to acoustic-phonetic features (e.g. voiced/unvoiced, the location of phoneme within a word). Each TDNN is trained independently, and then they are integrated hierarchically into a whole speech recognition system. In this study, we have experimented Korean plosives with phoneme-based recognition system and allophone-based recognition system. Experimental results show that allophone-based recognition is much less affected by the coarticulation.

• 한국음향학회지
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• 제14권3호
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• pp.97-104
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• 1995

본 논문은 음성 다이얼링 시스템 구현을 위한 연속 숫자음 인식에 관한 연구로써, 구문 분석을 이용한 One-Stage DP에 의한 음성 인식 방법을 제안하다. 인식 실험을 위해 우선 구간 구분화 알고리즘을 이용하여 DMS (Dynamic Multi-SEction) 모델을 만들며, 제안된 구문 분석을 이용한 One-Stage DP 방법으로 실험 대ㅛ상의 연속 숫자음 데이터를 인식하게 하였다. 본 연구에서는 8명의 ㅣ남성 화자에 의해 2-3번 발음도니 21종의 7자리의 연속 숫자음이 사용되었고, 기존의 One-Stage DP와 제안된 구문 분석을 이용한 One-Stage DP 음성 인식 알고리즘을 사용해서 화자 종속과 화자 독립 실험을 실험실 환경에서 수행하였다. 인식 실험 결과, 기존의 방법보다 제안된 방법이 인식률이 좋은 것으로 나타났으며, 제안된 방법에서는 화자 종속과 화자 독립 실험에서 각각 약 91.7%, 89.7%로 나타났다.