• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.1
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• pp.101-107
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• 2011

The background noises and distortions by channel are major factors that disturb the practical use of speech recognition. Usually, noise reduce the performance of speech recognition system DSR(Distributed Speech Recognition) based speech recognition also bas difficulty of improving performance for this reason. Therefore, to improve DSR-based speech recognition under noisy environment, this paper proposes a method which detects accurate speech region to extract accurate features. The proposed method distinguish speech and noise by using entropy and detection of spectral energy of speech. The speech detection by the spectral energy of speech shows good performance under relatively high SNR(SNR 15dB). But when the noise environment varies, the threshold between speech and noise also varies, and speech detection performance reduces under low SNR(SNR 0dB) environment. The proposed method uses the spectral entropy and harmonics of speech for better speech detection. Also, the performance of AFE is increased by precise speech detections. According to the result of experiment, the proposed method shows better recognition performance under noise environment.

• Annual Conference of KIPS
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• 2000.10a
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• pp.251-254
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• 2000

화자 독립 인식은 훈련 화자와 시험 화자의 차이로 인해 화자 종속의 경우보다 인식률이 떨어진다. 따라서, 인식률을 향상시키기 위해 화자 독립 모델을 화자에 적응시킬 필요가 있다. 본 논문에서는 효과적인 적응 방법인 MLLR(Maximum Likelihood Linear Regression) 적응 방법을 한국어 음성에 적용하여 적응 성능을 향상시켰고, 온라인 상에서 적용 가능하도록 증가 적응 방법을 이용하였다. PBW 445 음성 데이타베이스에 대한 실험 결과, 400개의 적응 데이터를 사용하였을 때, 제안한 방법이 기존의 화자 독립 시스템보다 7.02% 향상된 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.48-52
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• 1999

In this paper, we describe a Voice Command Web Browser using a variable vocabulary word recognizer that can do Internet surfing with Korean speech recognition on the Web. The feature of this browser is that it can handle the links and menus of the web browser by speech. Therefore, we can use speech interface together with mouse for web browsing. To recognize the recognition candidates dynamically changing according to Web pages, we use the variable vocabulary word recognizer. The recognizer was trained using POW (Phonetically Optimized Words) 3,848 words. So that it can recognize new words which did not exist in training data. The preliminary test results showed that the performance of speaker-independent and vocabulary-independent recognition is 93.8% for 32 Korean words. The Voice Command Web Browser was developed on windows 95/NT using Netscape Navigator and reflected usability test results in order to offer easy interface to users unfamiliar with speech interface. In on-line experiment of speaker-independent and environment-independent situation, Voice Command Web Browser showed recognition accuracy of 90%.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2007.11a
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• pp.443-446
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• 2007

음성신호는 잡음 또는 전송 채널의 특성에 의하여 왜곡되고, 왜곡된 음성은 음성인식 및 화자인식의 성능을 크게 저하시킨다. 이러한 문제점을 극복하기 위해 본 논문에서는 Gaussian mixture model (GMM)에 적용된 신호대잡음비 (SNR)기반 신뢰도 가중 기법[1][2]을 Hidden Markov model(HMM) 디코더에 변형하여 적용하였다. HMM 디코더 변형은 HMM 상태별 관측확률을 논문 [1]에서 제시된 신뢰도로 가중함으로써 이루어졌다. 제안한 방법의 성능을 확인하기 위해 ETRI에서 만든 한국어 화자인식용 휴대폰 음성 DB를 사용하여 문맥종속 화자식별 실험을 하였다. 실험결과 기존 방법에 비해 제안한 방법의 화자인식률이 크게 향상됨을 확인 할 수 있었다.

• Journal of Internet Computing and Services
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• v.3 no.4
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• pp.51-59
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• 2002

For the speech recognition, this paper proposes a improved new formant tracking algorithm The recognition data for the simulation on this paper are used with the Korean digit speech. The recognition rate of the improved algorithm for the Korean digit speech shows 91% for 300 digit speech The effectiveness of this research has been confirmed through recognition simulations.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.403-406
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• 1994

음성은 화자의 상태 및 주변 환경에 따라 그 특징이 다양하게 변화한다. 본 논문에서는 음성신호의 특징 파라미터로 널리 쓰이고 있는 mel-cepstrum에 대해, 단어내에서의 변화를 정규화함으로써 인식성능을 향상시키고자 하였다. mel-cepstrum이란 단어 전체에 대한 mel-cepstrum의 평균 값으로 normalize 시킨 것이다. 한국어 숫자음에 대한 인식 실험결과, 본 논문에서 제안한 정규화된 mel-cepstrum이 정규화되지 않은 mel-cepstrum에 비해 우수한 인식 성능을 나타내었다. 또한 잡음 환경하에서 비교 실험한 결과에서도 상대적으로 우수한 인식률을 보였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.13-16
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• 1999

이산 HMM을 이용한 음성인식을 할 때, 관측심볼은 훈련 데이터의 양자화과정을 수행하여 얻게 된다. 훈련 데이터는 선정된 몇몇 화자에 의해서 얻어지게 되는데, 이러한 이유로 인하여 충분하지 못한 훈련 데이터가 얻어지므로, 관측 심볼에 따라 출력확률분포값이 영(zero)이나, 거의 영에 가까운 값을 가지게 된다. 이러한 요인은 인식률의 저하를 초래하므로, 본 논문에서는 fuzzy smoothing 기법을 채택하여, 출력확률분포값이 영(zero)의 값을 가지는 것을 방지하여, 새로 구해진 파라메터로 인식실험을 하였다. Smoothing과정을 수행한 후의 인식률이 smoothing을 하진 않은 인식율에 비해 평균 $1.46\%$ 향상되었다.

• The Journal of the Acoustical Society of Korea
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• v.41 no.6
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• pp.655-661
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• 2022

Since speech impairment is prevalent in patients with Parkinson's disease (PD), speech recognition systems suitable for these patients are needed. In this paper, we propose a speech recognition system that effectively recognizes the speech of patients with PD. The speech recognition system is firstly pre-trained with the Globalformer using the speech data from healthy people, and then fine-tuned using relatively small amount of speech data from the patient with PD. For this analysis, we used the speech dataset of healthy people built by AI hub and that of patients with PD collected at Inha University Hospital. As a result of the experiment, the proposed speech recognition system recognized the speech of patients with PD with Character Error Rate (CER) of 22.15 %, which was a better result compared to other methods.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.3
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• pp.465-470
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• 2021

A robust speech recognition technology is required that does not degrade the performance of speech recognition and the quality of the speech when speech recognition is performed in an actual environment of the speech mixed with noise. With the development of such speech recognition technology, it is necessary to develop an application that achieves stable and high speech recognition rate even in a noisy environment similar to the human speech spectrum. Therefore, this paper proposes a speech enhancement algorithm that processes a noise suppression based on the MMSA-STSA estimation algorithm, which is a short-time spectral amplitude method based on the error of the least mean square. This algorithm is an effective nonlinear speech enhancement algorithm based on a single channel input and has high noise suppression performance. Moreover this algorithm is a technique that reduces the amount of distortion of the speech based on the statistical model of the speech. In this experiment, in order to verify the effectiveness of the MMSA-STSA estimation algorithm, the effectiveness of the proposed algorithm is verified by comparing the input speech waveform and the output speech waveform.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.6 no.3
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• pp.77-86
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• 2007

Recently, systems based on speech recognition interface such as telematics terminals are being developed. However, many errors still exist in speech recognition and then studies about error correction are actively conducting. This paper proposes an error correction in post-processing of the speech recognition based on features of Korean phoneme. To support this algorithm, we used the phoneme similarity considering features of Korean phoneme. The phoneme similarity, which is utilized in this paper, rams data by mono-phoneme, and uses MFCC and LPC to extract feature in each Korean phoneme. In addition, the phoneme similarity uses a Bhattacharrya distance measure to get the similarity between one phoneme and the other. By using the phoneme similarity, the error of eo-jeol that may not be morphologically analyzed could be corrected. Also, the syllable recovery and morphological analysis are performed again. The results of the experiment show the improvement of 7.5% and 5.3% for each of MFCC and LPC.