• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.57-60
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• 2004

본 논문에서는 음성정보기술산업지원센터(SiTEC)에서 현재 배포중인 고소음 환경 음성 DB를 이용하여 air-conducted 음성과 body-conducted 음성의 인식 성능을 비교 실험하였다. 소음 환경에서 일반적인 마이크로폰으로부터 수집된 air-conducted 음성은 잡음의 영향을 받기 쉬우며 이는 인식률을 저하시킨다. 반면에 진동 픽업 마이크로폰에서 수집된 body-conducted 음성은 소음에 보다 강인한 특성을 보인다. 이러한 특성에 근거하여 소음 환경에서 일반 다이나믹 마이크로폰 음성에 음질 개선 방법과 채널 보상 방법을 적용한 인식 결과와 3종류의 진동 픽업 마이크로폰에서 수집된 음성과의 인식 성능을 비교 분석하여 body-conducted 음성 인식 시스템의 환용 가능성을 살펴보았다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.3
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• pp.451-458
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• 2001

In this paper, we propose speech control system for a various control device in the car with real time control speech. A real time speech control system is detected start-end points from speech data processing by A/D conversion, and recognize by one pass dynamic programming method. The results displays a monitor, and transports control data to control interfaces. The HMM model is modeled by a continuous control speech consists of control speech and digit speech for controlling of a various control device in the car The recognition rates is an average 97.3% in case of word & control speech, and is an average 96.3% in case of digit speech.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.59-68
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• 1994

This paper describes the performance assessment test and analysis of test results on a Korean speech recognizer which recognizes Lombard effect received speech in noisy environment, as a basic performance assessment research. In the assessement test, standard speech data were first manipulated close to speech uttered in a noisy environment, and then performance assessment tests were carried out along with the assessment items (the type of noise, SNR) in two ways-one with Lombard effect received speech(LES), the other with not received(NLES). As a result, when 90% of recognition rate is set to be a recognition limit, it was achieved at 10dB SNR point with LES, while at 30dB with NLES. This 20dB of SNR difference indicates Lombard effect should be considered in real world assessment test. The type of noises didn't affect performance of recognizers in out tests. ANOVA analysis, in evaluating several kinds of recognizers, showed every assessment item affecting the recognition performance could be quantified.

• Journal of Korea Multimedia Society
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• v.16 no.8
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• pp.905-913
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• 2013

The Parallel Model Compensation Using Frequency-variant: FV-PMC for noise-robust speech recognition is a method to classify the noises, which are expected to be intermixed with input speech when recognized, into several groups of noises by setting average frequency variant as a threshold value; and to recognize the noises depending on the classified groups. This demonstrates the excellent performance considering noisy speech categorized as good using the standard threshold value. However, it also holds a problem to decrease the average speech recognition rate with regard to unclassified noisy speech, for it conducts the process of speech recognition, combined with noiseless model as in the existing PMC. To solve this problem, this paper suggests a enhanced method of recognition to prevent the unclassified through improving the extent of rating scales with use of maximum likelihood so that the noise groups, including input noisy speech, can be classified into more specific groups, which leads to improvement of the recognition rate. The findings from recognition experiments using Aurora 2.0 database showed the improved results compared with those from the method of the previous FV-PMC.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.41 no.3
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• pp.67-77
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• 2004

The recent research has been focused on fusion of audio and visual features for reliable speech recognition in noisy environments. In this paper, we propose a neural network based model of robust speech recognition by integrating audio, visual, and contextual information. Bimodal Neural Network(BMNN) is a multi-layer perception of 4 layers, each of which performs a certain level of abstraction of input features. In BMNN the third layer combines audio md visual features of speech to compensate loss of audio information caused by noise. In order to improve the accuracy of speech recognition in noisy environments, we also propose a post-processing based on contextual information which are sequential patterns of words spoken by a user. Our experimental results show that our model outperforms any single mode models. Particularly, when we use the contextual information, we can obtain over 90% recognition accuracy even in noisy environments, which is a significant improvement compared with the state of art in speech recognition. Our research demonstrates that diverse sources of information need to be integrated to improve the accuracy of speech recognition particularly in noisy environments.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.41-44
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• 2004

본 논문에서는 한국어에만 나타나는 경음에 대하여 음운학적, 음향학적 특성을 고찰하고 이를 기반으로 음성인식 실험을 수행한 후 그 결과를 분석하였다. 음성인식 실험을 위하여 입력 음성을 48개의 유사음소단위 (PLU; Phoneme Likely Unit)로 레이블링을 한 후 각각의 음소군에 대하여 LPC (Liner Predictive Coding) 분해능을 증가시키면서 음소인식 및 단어인식 실험을 수행하였다. 그 결과, 음소 인식 실험에서 경음군의 인식률이 가장 낮게 나타나 경음에 대한 분석이 보다 많이 필요함을 알 수 있었다. 또한 PLC의 분해 차원이 23차 일 때 경음과 전체 음소 인식률이 각각 $34.11\%,\;46.1\%$로 나타나 가장 양호함을 알 수 있었으며 단어인식 실험에서도 LPC 23차와 25차 일 때 $81.68\%,\;81.87\%$로 인식률이 가장 좋음을 알 수 있었다. 이상의 실험 결과에서 한국어의 경음은 전체 시스템의 인식 성능과 밀접한 관계가 있음을 알 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9C
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• pp.907-912
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• 2005

An efficient method for calculation of observation probability in CDHMM(Continous Density Hidden Markov Model) is proposed in this paper. the proposed algorithm, called FCOP(Fast Computation of Observation Probability), approximate obsewation probabilities in CDHMM by eliminating insignificant PDFs(Probability Density Functions) and reduces the computational load. When applied to a speech recognition system, the proposed FCOP algorithm can reduce the instruction cycles by $20\%-30\%$ and can also increase the recognition speed about $30\%$ while minimizing the loss in its recognition rate. When implemented on a practical cellular phone, the FCOP algorithm can increase its recognition speed about $30\%$ while suffering $0.2\%$ loss in recognition rate.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.41-50
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• 1994

This paper proposes a DMS(Dynamic Multi-Section) model based on the information of the similar features in word pattern. This model represents each word as a time series of several sections and each section implies duration time information and typical feature vectors. The procedure to make a model in the word pattern is that typical feature vector and duration time information are reflected in the distance, when matching between word pattern and model is repeated. As the result of it, the accumulated distance by matching is to be minimized.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.6 no.3
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• pp.87-94
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• 2007

In this paper, the performance of the speech recognition and speaker adaptation methods are studied. The speech recognition using energy state and VQ(Vector Quantization) is suggested and the speaker adaptation methods(Maximum a posteriori probability estimation, linear specrum estimation) are considered. The experimental results show that recognition ration using energy state is 2-3 % better than that of general VQ.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.281-284
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• 2001

In this paper, we present a car number retrieving system using speech recogntion and speech synthesis for PDA. This system consist of 4-digit numbers and command speech recognition as well its speech synthesis. Experiment results showed 4-digit numbers recognition rate 97% and commands recognition 99% through speaker-independent method.