• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.97-103
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• 2001

The speaker recognition method using DTW algorithm has the problem that is reducing the performance of the speaker recognition system as the time variation. So there are many proposed algorithms to solve these problems. This paper proposes the new method If make the reference pattern that is acceptable to intra-speaker variation by reference pattern normalization. And to avoid reducing performance of speaker recognition system, we use the modified reference pattern to recognize the system user. The used methods in this paper are VQ and DTW. As the result of simulation we can obtain the $97.5\%$ of recognition accuracy rate.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.183-186
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• 2002

최근 음성인식의 인식 단위로서 문맥의존 음향 모델이 널리 사용되고 있다. 이는 음소의 음향학적 특징, 즉 선행 및 후행음소에 의한 중심 음소의 변이음 모델이 문맥독립 모델보다 좀 더 정확하게 모델링 될 수 있기 때문이다. 하지만 강건한 문맥의존 음향 모델을 작성하기 위해서는 모델 파라미터의 병합(tying)과 미지의 문맥(unseen context)의 처리를 위한 좀더 정교한 해결 방법이 필요하다. 따라서 본 논문에서는 이점을 고려하여 음향학적 특징과 언어학적 특징을 결합하여 상태 분할을 수행할 수 있도록 SSS(Successive State Splitting) 알고리즘의 문맥 방향 상태 분할에 음소결정트리를 접목한 HM-Net(Hidden Markov Network) 구조 결정법을 도입하였다. 또한 HM-Net은 연속적인 상태 분할에 의해 한국어에서 많이 발생하는 변이음들을 효과적으로 모델링 할 수 있다는 점을 고려하여 본 연구실에서 기존에 사용하던 48 유사음소 단위에서 문맥의존 음향 모델 작성에 불필요한 변이음을 제거하여 39 유사음소 단위를 재 정의하였다. 도입한 방법과 새로 정의한 유사음소 단위의 유효성을 확인하기 위해 고립 단어, 4연속 숫자음, 연속 음성인식에 대해 인식 실험을 수행한 결과, 모든 실험에서 재 정의한 39 유사음소 단위가 문맥종속형 HM-Net 음향모델을 이용한 한국어 음성인식에 효과적임을 확인할 수 있었다. 특히 연속 음성인식 실험의 경우, 기존의 48 유사음소 단위보다 평균 $15.08\%$의 인식률 향상이 있었다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.543-551
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• 2002

In this paper, we present an efficient decoding method that performs in real time for 20k word continuous speech recognition task. Basic search method is a one-pass Viterbi decoder on the search space constructed from the novel language model network. With the consistent search space representation derived from various language models by the LM network, we incorporate basic pruning strategies, from which tokens alive constitute a dynamic search space. To facilitate post-processing, it produces a word graph and a N-best list subsequently. The decoder is tested on the database of 20k words and evaluated with respect to accuracy and RTF.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.170-178
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• 2012

In the paper, phonetic decision tree of the triphone unit was built for the phoneme-based speech recognition of 640 stations which run by the Korail. The clustering rate was determined by Pearson and Regression analysis to decide threshold used in node splitting. Using the determined the clustering rate, thresholds are automatically decided by the threshold value according to the average clustering rate. In the recognition experiments for verifying the proposed method, the performance improved 1.4~2.3 % absolutely than that of the baseline system.

• Journal of Digital Contents Society
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• v.10 no.1
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• pp.121-127
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• 2009

The automotive ECU(Electronic Control Unit) becomes more complicated and is demanding many functions. For example, many automobile companies are developing driver convenience systems such as power window switch, LCM(Light Control Module), mirror control system, seat memory. In addition, many researches and developments for DIS(Driver Information System) are in progress. It is dangerous to operate such systems in driving. In this paper, we implement the speech recognition system which controls the car convenience system using speech, and apply the preprocessing filter to improve the speech recognition rate in car noise environment. As a result, we get the good speech recognition rate in car noise environment.

• 전자공학회논문지 IE
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• v.43 no.2
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• pp.22-29
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• 2006

Recently the use of whispered speech has increased due to mobile phone and the necessity of whispered speech recognition is increasing. So various feature vectors, which are mainly used for speech recognition, are applied to their HMMs, normal speech models, whispered speech models, and integrated models with normal speech and whispered speech so as to find out suitable recognition system for whispered speech. The experimental results of recognition test show that the recognition rate of whispered speech applied to normal speech models is too low to be used in practical applications, but separate whispered speech models recognize whispered speech with the highest rates at least 85%. And also integrated models with normal speech and whispered speech score acceptable recognition rate but more study is needed to increase recognition rate. MFCE and PLCC feature vectors score higher recognition rate when applied to separate whispered speech models, but PLCC is the best when a lied to integrated models with normal speech and whispered speech.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.85-89
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• 1996

In this paper, transition constrained Hidden Markov Model(HMM) in which the transition between states occur only within prescribed time slot is proposed and the performance is evaluated in the noisy environment. The transition constrained HMM can explicitly limit the state durations and accurately de scribe the temporal structure of speech signal simply and efficiently. The transition constrained HMM is not only superior to the conventional HMM but also require much less computation time. In order to evaluate the performance of the transition constrained HMM, speaker independent isolated word recognition experiments were conducted using semi-continuous HMM with the noisy speech for 20, 10, 0 dB SNR. Experiment results show that the proposed method is robust to the environmental noise. The 81.08% and 75.36% word recognition rates for conventional HMM was increased by 7.31% and 10.35%, respectively, by using transition constrained HMM when two kinds of noises are added with 10dB SNR.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.61-64
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• 2001

본 논문에서는 class간의 변별력을 증가시키기 위한 유용한 방법인 LDA(Linear Discriminant Analysis)를 사용하여 한국어 숫자음간의 변별력을 높여 연결숫자 인식성능을 높이고자 하였다. 한국어 숫자음은 모두 단음절이며 또한 혼동가능성이 높은 숫자쌍이 존재하여 이것이 전체 인식률을 저하시킨다. LDA를 사용한 경우 숫자열 오인식률이 $8\%$ 감소하였다. 그리고, 음성특징 벡터의 차수를 감소시키고 LDA 사용전 보다 약간의 인식률 증가를 보였다. 그러나, 선형적인 방법으로 분리가 불가능한 class들의 분포가 존재할 때는 LDA를 사용하여도 변별력 향상은 기대하기 어렵다. 이와 같은 분포의 class사이의 변별력을 증가시키기 위해 between-scatter covariance matrix를 구할 때 class 사이에 혼동가능성 정도를 나타내는 weighting factor를 적용하였으며, 그 결과 숫자열 오인식률이 LDA 사용전보다 $9.7\%$ 감소하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.179-183
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• 1995

DTW를 사용한 텍스트종속 화자확인의 성능향상에 관한 것으로, 화자인식의 근본적인 난점인 화자 정보 추출의 어려움, 사칭자의 거부, 시간 변화에 따른 인식률 저하 등을 해결하고자 하였다. 먼저 기존의 DTW 방식을 유지하면서 DTW의 단점이라 할 수 있는 과다한 계산량과 발성 습관과 시간 변화에 따른 음성왜곡을 개선하기 위하여 기준 패턴에 통계적 의미를 도입한 대표 평균치 패턴을 사용하였다. 가중 켑스트럼은 화자별로 유용한 켑스트럼 차수를 구하여, 그 차수에 가중치를 두는 것으로 본 실험에서는 F-ratio를 사용하여 구하였다. 실험결과 대표 평균치 패턴과 F-ratio를 사용할 경우 인식률이 각각 약 3~4% 향상되었다.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.9
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• pp.103-112
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• 1999

In this paper, a speaker-independent isolated world recognition system for a car navigation system is implemented using a general digital signal processor. This paper presents a method combining SNR normalization with RAS as a noise processing method. The semi-continuous hidden markov model is adopted and TMS320C31 is used in implementing the real-time system. Recognition word set is composed of 69 command words for a car navigation system. Experimental results showed that the recognition performance has a maximum of 93.62% in case of a combination of SNR normalization and spectral subtraction, and the performance improvement rate of the system is 3.69%, Presented noise processing method showed good speech recognition performance in 5dB SNR in car environment.