• 한국정보통신학회논문지
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• 제18권7호
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• pp.1740-1748
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• 2014

본 논문에서는 구개인두부전증 (VPI) 환자 발음과 정상인의 모의 발음에 대한 듣기 평가와 음향 분석을 실시한다. 본 연구를 위해 음성 데이터 수집을 위해 50개의 단어, 모음 및 단음절로 이루어진 발음 목록을 설정한다. 듣기 평가실험의 편의를 위해 웹 기반의 듣기 평가 시스템을 구축한다. 듣기 평가 결과는 실제 VPI 환자의 발음에 대한 오인식 경향과 모의 발음의 오인식 경향이 유사함을 나타낸다. 이러한 유사성은 모음의 포먼트 위치와 자음의 스펙트럼의 비교를 통해서도 확인할 수 있다. 실험 결과는 본 연구에서 사용한 정상인의 VPI 모의 발화 기법이 실제 환자의 음성을 비교적 효과적으로 모의하는 것을 반영하는 결과이다. 향후 VPI 환자의 음성 인식 과정에서 정상인의 모의 발화음성 데이터를 음향 모델의 적응 기법과 같은 분야에 유용하게 사용할 수 있을 것으로 기대한다.

• 대한의용생체공학회:의공학회지
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• 제28권2호
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• pp.205-211
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• 2007

A novel envelope extraction algorithm for speech processor of cochlear implants, called adaptation algorithm, was developed which is based on a adaptation effect of the inner hair cell(IHC)/auditory nerve(AN) synapse. We achieved acoustic simulation and hearing experiments with 12 normal hearing persons to compare this adaptation algorithm with existent standard envelope extraction method. The results shows that speech processing strategy using adaptation algorithm showed significant improvements in speech recognition rate under most channel/noise condition, compared to conventional strategy We verified that the proposed adaptation algorithm may yield better speech perception under considerable amount of noise, compared to the conventional speech processing strategy.

• 음성과학
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• 제11권4호
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• pp.53-66
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• 2004

The nonlinear speech enhancement method with interactive parallel-extended Kalman filter is applied to speech contaminated by additive white noise. To represent the nonlinear and nonstationary nature of speech. we assume that speech is the output of a nonlinear prediction HMM (NPHMM) combining both neural network and HMM. The NPHMM is a nonlinear autoregressive process whose time-varying parameters are controlled by a hidden Markov chain. The simulation results shows that the proposed method offers better performance gains relative to the previous results [6] with slightly increased complexity.

• 한국통신학회:학술대회논문집
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• 한국통신학회 1984년도 춘계학술발표회논문집
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• pp.18-21
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• 1984

In this paper, the tree coding using the (M, L) multi-path search algorithm has teen investigated. A hybrid adaptation scheme which employs a block adaptation as well as a sequential dadptation is described for application in quantization and compression of speech signals. Simulation results with the gybrid adaptation scheme indicate that a relatively good speech quality can be obtained at rate about 8Kbps. All necessary parameters such as MlL and filter-order were found from simulation and these parameters turned out to be a good compromise between the complexity and overall performance.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 추계학술대회 논문집 학회본부 B
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• pp.700-702
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• 1999

The protection of data which we want to keep secret from invalid users has become a main topic nowadays. This paper introduces a encryption scheme for protecting speech signals from eavesdropping. The proposed encryption scheme adopts a secure voice cryptographic algorithm based on the scrambling in frequency band. In order to improve the conventional speech signal encryption scheme, we have randomly permuted DCT coefficients of speech signal. Simulation results are included to show the performance of the proposed algorithm for secure transmission of speech signals.

• 한국통신학회논문지
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• 제21권5호
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2005년도 추계 학술대회 발표논문집
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• pp.121-124
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• 2005

In this paper, we propose a sound localization algorithm for two simultaneous speakers. Because speech is wide-band signal, there are many frequency sub-bands in that two speech sounds are mixed. However, in some sub-bands, one speech sound is more dominant than other sounds. In such sub-bands, dominant speech sounds are little interfered by other speech or noise. In speech sounds, overtones of fundamental frequency have large amplitude, and that are called 'Harmonic structure of speech'. Sub-bands inharmonic structure are more likely dominant. Therefore, the proposed localization algorithm is based on harmonic structure of each speakers. At first, sub-bands that belong to harmonic structure of each speech signal are selected. And then, two speakers are localized using selected sub-bands. The result of simulation shows that localization using selected sub-bands are more efficient and precise than localization methods using all sub-bands.

• 전기학회논문지
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• 제62권12호
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• pp.1750-1754
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• 2013

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

• 대한음성학회지:말소리
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• 제51호
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• pp.137-149
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• 2004

A weather forecast service with speech recognition is described. This service allows users to get the weather information of all the cities by saying the city names with just one phone call, which was not provided in the previous weather forecast service. Speech recognition is implemented in the intelligent peripheral (IP) of the advanced intelligent network (AIN). The AIN is a telephone network architecture that separates service logic from switching equipment, allowing new services to be added without having to redesign switches to support new services. Experiments in speech recognition show that the recognition accuracy is 90.06% for the general users' speech database. For the laboratory members' speech database, the accuracies are 95.04% and 93.81%, respectively in simulation and in the test on the developed system.

• 한국음향학회지
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• 제23권8호
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• pp.597-601
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• 2004

본 논문에서는 CDMA이동통신 환경에서의 음성인식을 위한 왜곡음성신호의 전처리-지부방법을 소개한다. 먼저, CDMA이동통신 채널에서의 왜곡된 음성신호를 분석하고 분석된 매커니즘을 바탕으로 채널에 의해 왜곡된 음성신호를 음성의 준주기성을 바탕으로 하여 거부하는 알고리즘을 제안한다. 실험을 통해 제안된 전처리-거부방법이 적은 계산량을 가지고 음성인식에 적용되어 효과적으로 CDMA에 환경에서 채널왜곡된 음성신호를 거부-할 수 있음을 알 수 있었다.