• 한국멀티미디어학회논문지
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• 제18권4호
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• pp.437-442
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• 2015

Voice activity detection is very important process that voice activity separated form noisy speech signal for speech enhance. Over the past few years, many studies have been made on voice activity detection, but it has poor performance in low signal to noise ratio environment or fickle noise such as car noise. In this paper, it proposed new voice activity detection algorithm using ensemble variance based on wavelet band entropy and soft thresholding method. We conduct a survey in a lot of signal to noise ratio environment of car noise to evaluate performance of the proposed algorithm and confirmed performance of the proposed algorithm.

• ETRI Journal
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• 제23권1호
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• pp.16-22
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• 2001

Efficient quantization methods of the line spectrum pairs (LSP) which have good performances, low complexity and memory are proposed. The adaptive quantization range method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. As the maximum quantization range of each LSP parameter is varied adaptively on the quantized value of the previous order's LSP parameter, efficient quantization methods can be obtained. The proposed scalar quantization algorithm needs 31 bits/frame, which is 3 bits less per frame than in the conventional scalar quantization method with interframe prediction to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1 dB using 26 bits/frame. The performances of proposed quantization methods are also evaluated in the transmission errors.

• 대한음성학회지:말소리
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• 제66호
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• pp.73-86
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• 2008

Beamforming is one of the spatial filtering techniques which extract only desired signals from noisy environments using microphone arrays. Fixed beamforming is a simple concept and easy to implement. However, it does not show good performance in real noisy conditions. As an adaptive beamforming, Frost algorithm can be a good candidate. It uses the concept of the linearly constrained minimum variance (LCMV) algorithm. The difference between the Frost and the LCMV algorithm is the error correction scheme which is very effective feature in the aspect of performance. In this paper, as quadrature mirror filtering (QMF)-based filterbank is utilized as the pre-processing of the Frost beamformning, the filter length and the learning rate of each band is optimized to improve the performance. The performance is measured by the signal-to-noise ratio (SNR) and the Bark's scale spectral distortion (BSD).

• 한국통신학회논문지
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• 제18권11호
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• pp.1677-1688
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• 1993

호핑 필터를 이용한 이차원 진폭 스크램블링 알고리즘은 기존의 음성 보호 방식의 단점을 개선시킬 수 있는 아날로그 음성 신호에 있어서 강력한 보호 방식이다. 본 논문에서는 이차원 진폭 스크램블링 알고리즘의 최대 단점인 동기 문제를 해결하기 위하여 variable delay를 이용한 알고리즘을 제안하였다. 또한 PAM 신호를 가우시안 집음이 존재하는 채널로 전송하였을 때 수신단에서는 복원된 음성 신호의 왜곡을 분석함으로써 최적의 보호 알고리즘 및 최적의 SNR 값을 시뮬레이션을 이용하여 나타내었다.

• 전자공학회논문지B
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• 제30B권2호
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• pp.51-58
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• 1993

This paper describes the development of a speaker-dependent isolated word recognizer as applied to voice dialing in a car noise environment. for this purpose, several methods to improve performance under such condition are evaluated using database collected in a small car moving at 100km/h The main features of the recognizer are as follow: The endpoint detection error can be reduced by using the magnitude of the signal which is inverse filtered by the AR model of the background noise, and it can be compensated by using variants of the DTW algorithm. To remove the noise, an autocorrelation subtraction method is used with the constraint that residual energy obtainable by linear predictive analysis should be positive. By using the noise rubust distance measure, distortion of the feature vector is minimized. The speech recognizer is implemented using the Motorola DSP56001(24-bit general purpose digital signal processor). The recognition database is composed of 50 Korean names spoken by 3 male speakers. The recognition error rate of the system is reduced to 4.3% using a single reference pattern for each word and 1.5% using 2 reference patterns for each word.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2005년도 춘계 학술대회 발표논문집
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• pp.49-52
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• 2005

Cepstral Mean Subtraction (CMS) makes effectively compensation for a channel distortion, but there are some shortcomings such as distortions of feature parameters, waiting for the whole speech sentence. By assuming that the silence parts have the channel characteristics, we consider the channel normalization using subtraction of cepstral means which are only obtained in the silence areas. If the considered techniques are successfully used for the channel compensation, the proposed method can be used for real time processing environments or time important areas. In the experiment result, however, the performance of our method is not good as CMS technique. From the analysis of the results, we found potentiality of the proposed method and will try to find the technique reducing the gap between CMS and ours method.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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• pp.274-277
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• 2007

The purpose of this study is 1) to describe the phoneme inventories of cochlear implant(CI) children and 2) to describe their utterances using narrow phonetic transcription method. All the subjects had more than 2 year-experience with CI and showed more than 87% open-set sentence perception abilities. Average consonant accuracy was 81.36% and it was improved up to 87.41% when distortion errors were not counted. They showed different error patterns from hearing aid users. The prominent error pattern was weakening of consonants.

• 대한전자공학회논문지
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• 제12권3호
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• pp.30-36
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• 1975

TDM 회선을증가하기 위한 두가지 방식-주기적 sample skipping방법과 표본화주파수 축소 방법-에 대한 불특성을 해석비교하였다. 신호는 통계적으로 stationary인 random신호로 가정했으며, 선로의 잡음과 각랑 상호 간의 간섭효과는 고려하지 않았다. 음성신호에 대한 구체적 비교 결과, 주기회 sample skipping방법이 실제적 설계조건에서 훌륭한 선택이 될 수 있음을 보였다. Distortions are analyzed and compared for two TDM channel expansion methods- periodic sample skipping and sampling frequency reduction. Signal is assumed to be stationary random signal with zero·mean. Channel noise and interference are not considered in the analysis. For speech signal, it is shown that the periodic sample skipping method could be a better choice under practical design constraints.

• 음성과학
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• 제11권4호
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• pp.163-171
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• 2004

The primary goal of the present study was to explore more detailed evidence for the influence of aging and gender effects on the capability of Korean healthy, ears to generate DPOAEs. DPOAEs were examined in series of human subjects, with clinically nonnal hearing, ranging in age from 10 to 65 years. All 60 Koreans were divided into 6 age groups. Each age group included 10 participants, 5 females and 5 males. The gender effects on the difference between the absolute amplitude and noise floor value in DPOAEs did not exist. The difference increased as the frequency increased. The aging effects on the difference between the absolute amplitude and noise floor value in DPOAEs did exist. The difference increased as the frequency increased but orderly age effects could not be found. The principle finding was that, when compared between emissions in young and old ears, DPOAEs accurately tracked the systematic deterioration of high-frequency hearing in aging individuals. Such results support the need to reestablish the criterion for interpretation of DPOAEs in the elderly.

• 디지털융복합연구
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• 제12권10호
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• pp.265-270
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• 2014

기존의 음성 인식 알고리즘은 어휘들 간의 순서가 정해져 있지 않으며, 음성 인식 환경 변화에 따른 잡음으로 인한 음성 검출이 정확하지 못한 단점을 가지며, 검색 시스템은 키워드의 의미가 다양하여 정확한 정보를 인지하지 못한다. 본 연구에서는 사건 기반 시맨틱 온톨로지 추론 모델을 제안하였으며, 제안된 시스템에서 음성 인식 특징을 추출하기 위해 ERB 필터를 이용하여 특징 추출하는 모델을 구축하였다. 제안된 모델은 성능 평가를 위해 지하철역, 지하철 잡음을 사용하였고 잡음 환경의 SNR -10dB, -5dB 신호에서 잡음 제거를 수행하여 왜곡도를 측정한 결과 2.17dB, 1.31dB의 성능이 향상됨을 확인하였다.