• Journal of Korea Multimedia Society
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• v.10 no.11
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• pp.1530-1537
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• 2007

In this paper, we implemented a speech emotion recognition system that can distinguish human emotional states from recorded speech captured by a single microphone and classify them into four categories: neutrality, happiness, sadness and anger. In general, a speech recorded with a microphone contains background noises due to the speaker environment and the microphone characteristic, which can result in serious system performance degradation. In order to minimize the effect of these noises and to improve the system performance, a MA(Moving Average) filter with a relatively simple structure and low computational complexity was adopted. Then a SFS(Sequential Forward Selection) feature optimization method was implemented to further improve and stabilize the system performance. For speech emotion classification, a SVM pattern classifier is used. The experimental results indicate the emotional classification performance around 65% in the computer simulation and 62% on the ARM platform.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.19-22
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• 2000

어휘독립 음성인식이란 음향학적 모델 훈련에 사용하지 않은 어휘들을 인식하는 것이다. 단어모델을 이용한 어휘독립 음성인식 시스템은 발음표기로 변환된 인식대상어휘에 대하여 문맥 종속형 부단어(context dependent subword) 단위로 훈련된 모델을 연결하여 단어 모델을 만들고 이 단어 모델로 인식을 수행한다. 이러한 시스템의 경우 훈련과정에서 나타나지 않는 문맥 종속형 부단어가 인식대상어휘에서 나타나게 되고, 따라서 정확한 단어모델을 구성할 수 없다는 문제점이 있다 본 논문에서는 문맥 종속형 부단어 구분의 계층화를 통한 back-off 선택 방법을 이용하여 새롭게 나타난 문맥 종속형 부단어 대신 연결될 부단어 모델을 찾아내는 방법을 제안한다 제안된 선택 방법은 새롭게 나타난 문맥 종속형 부단어를 포함하는 상위의 부단어를 찾아내는 방법이다. 실험 결과 10단어 세트에서 $97.5\%$ 50단어 세트에서$90.16\%$ 100 단어 세트에서 $82.08\%$의 인식률을 얻었다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.4
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• pp.991-998
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• 2010

In realization of wireless home network system using speech recognition in indoor voice recognition environment, background noise and reverberation are two main causes of digression in voice recognition system. In this study, the home network system resistant to reverberation and background noise using voice section detection method based on spectral entropy in indoor recognition environment is to be realized. Spectral subtraction can reduce the effect of reverberation and remove noise independent from voice signal by eliminating signal distorted by reverberation in spectrum. For effective spectral subtraction, the correct separation of voice section and silent section should be accompanied and for this, improvement of performance needs to be done, applying to voice section detection method based on entropy. In this study, experimental and indoor environment testing is carried out to figure out command recognition rate in indoor recognition environment. The test result shows that command recognition rate improved in static environment and reverberant room condition, using voice section detection method based on spectral entropy.

• The Journal of the Acoustical Society of Korea
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• v.38 no.6
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• pp.703-709
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• 2019

This paper proposes speech recognition systems employing Convolutional Neural Network (CNN) and Long Short Term Memory (LSTM) structures combined with Hidden Markov Moldel (HMM) to effectively recognize the speech of VeloPharyngeal Insufficiency (VPI) patients, and compares the recognition performance of the systems to the Gaussian Mixture Model (GMM-HMM) and fully-connected Deep Neural Network (DNNHMM) based speech recognition systems. In this paper, the initial model is trained using normal speakers' speech and simulated VPI speech is used for generating a prior model for speaker adaptation. For VPI speaker adaptation, selected layers are trained in the CNN-HMM based model, and dropout regulatory technique is applied in the LSTM-HMM based model, showing 3.68 % improvement in recognition accuracy. The experimental results demonstrate that the proposed LSTM-HMM-based speech recognition system is effective for VPI speech with small-sized speech data, compared to conventional GMM-HMM and fully-connected DNN-HMM system.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.199-210
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• 2002

In this paper, we carried out the study on speech recognition using the KM-Net topology design algorithm based on decision tree state-clustering to improve the performance of acoustic models in speech recognition. The Korean has many allophonic and grammatical rules compared to other languages, so we investigate the allophonic variations, which defined the Korean phonetics, and construct the phoneme question set for phonetic decision tree. The basic idea of the HM-Net topology design algorithm is that it has the basic structure of SSS (Successive State Splitting) algorithm and split again the states of the context-dependent acoustic models pre-constructed. That is, it have generated. the phonetic decision tree using the phoneme question sets each the state of models, and have iteratively trained the state sequence of the context-dependent acoustic models using the PDT-SSS (Phonetic Decision Tree-based SSS) algorithm. To verify the effectiveness of the above algorithm we carried out the speech recognition experiments for 452 words of center for Korean language Engineering (KLE452) and 200 sentences of air flight reservation task (YNU200). Experimental results show that the recognition accuracy has progressively improved according to the number of states variations after perform the splitting of states in the phoneme, word and continuous speech recognition experiments respectively. Through the experiments, we have got the average 71.5%, 99.2% of the phoneme, word recognition accuracy when the state number is 2,000, respectively and the average 91.6% of the continuous speech recognition accuracy when the state number is 800. Also we haute carried out the word recognition experiments using the HTK (HMM Too1kit) which is performed the state tying, compared to share the parameters of the HM-Net topology design algorithm. In word recognition experiments, the HM-Net topology design algorithm has an average of 4.0% higher recognition accuracy than the context-dependent acoustic models generated by the HTK implying the effectiveness of it.

• Proceedings of the KAIS Fall Conference
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• 2008.11a
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• pp.316-319
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• 2008

본 논문에서는 유아의 장난감에 RFID 시스템을 도입하여 유아가 흥미있어하는 장난감에 Tag를 삽입하고 Display 장치 부근에 RFID-Reader를 설치하여 장남감이 Reader 근처에 위치하게 되었을 때 장난감에 해당되는 학습정보를 Display 장치를 통하여 보여준다. 본 논문의 결과 유아가 장난감을 가지고 노는 동안에 영상, 음성 등의 콘텐츠를 Display 장치를 이용하여 보여주고 들려 줌으로써 자연스럽게 장난감의 정보를 학습할 수 있도록 하였다. RFID의 인식률과 인식거리를 감안하여 13.56MHz의 수동형 태그를 사용하였으며, 최대 인식거리는 20Cm에서 인식할 수 있도록 하였다. 실험은 RFID의 인식률과 콘텐츠의 자동인식 및 실행에 중점을 두었으며, RFID의 인식은 하나의 태그 인식에서는 99%이상의 성공을 하였고 두개이상의 태그에 대해서는 첫 번째 태그만을 인식할 수 있게 설계하여 콘텐츠를 실행하도록 하였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.10-16
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• 1999

In this paper, we have studied Continuous Speech Recognition System of Speaker Adaptation using MAPE (Maximum A Posteriori Probability Estimation) which can adapt any small amount of adaptation speech data. Speaker adaptation is performed by the method of MAPB after Concatenation training which is making sentence unit HMM linked by syllable unit HMM and Viterbi segmentation classifies speech data to be adaptation into segmentation of syllable unit data automatically without hand labelling. For car control speech the recognition rates of adaptation of HMM was 77.18% which is approximately 6% improvement over that of unadapted HMM.(in case of O(n)DP)

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.559-563
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• 2008

The purpose of this paper is to introduce a multimodal interface for mobile phones and to verify its feasibility. The multimodal interface integrates multiple input devices together including speech, keypad and motion. It can enhance the late and time for speech recognition, and shorten the menu depth.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2004.10a
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• pp.560-563
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• 2004

본 논문은 분노, 행복, 평정, 슬픔, 놀람 둥과 같은 인간의 감정상태를 인식하는 새로운 접근에 대해 설명한다. 이러한 시도는 이산길이를 포함하는 연속 은닉 마르코프 모델(HMM)을 사용함으로써 이루어진다. 이를 위해, 우선 입력음성신호로부터 감정의 특징 파라메타를 정의 한다. 본 연구에서는 피치 신호, 에너지, 그리고 각각의 미분계수 등의 운율 파라메타를 사용하고, HMM으로 훈련과정을 거친다. 또한, 화자적응을 위해서 최대 사후확률(MAP) 추정에 기초한 감정 모델이 이용된다. 실험 결과, 음성에서의 감정 인식률은 적응 샘플수의 증가에 따라 점차적으로 증가함을 보여준다.

• The Journal of the Acoustical Society of Korea
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• v.13 no.1E
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• pp.20-24
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• 1994

A new speech recognition model, DLSMM(Dynamically Localized Self-organizing Map Model) and its effective training algorithm are proposed in this paper. In DLSMM, temporal and spatial distortions of speech are efficiently normalized by dynamic programming technique and localized self-organizing maps, respectively. Experiments on Korean digits recognition have been carried out. DLSMM has smaller Experiments on Korean digits recognition have been carried out. DLSMM has smaller connections than predictive neural network models, but it has scored a little high recognition rate.