• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.58-62
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• 1999

This paper proposes use of acoustic parameters to improve the discriminability among digit models in Korean connected digit recognition. The proposed method used the logarithmic values of energy ratio between the predetermined frequency bands as additional feature parameters, based on the acoustic-phonetic knowledge. The results of our experiment show that the proposed method reduced the error rate by 46% in comparison with the baseline system. And incorporation of channel compensation technique in the proposed method yielded error reduction of about 69%.

• Proceedings of the Acoustical Society of Korea Conference
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• 1997.06a
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• pp.57-63
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• 1997

실린더 형태의 관은 산업체나 일반가정에서 자주 사용되어진다. 이러한 실린더 형태의 관들은 유체나 기체를 운반하는데 쓰이며, 운반과정에서 소음이 발생한다. 이러한 소음들의 해석은 우선실린더 방정식을유도하여야 한다. 음향학 분야에서 여러 가지 다른 형태의 실린더 운동 방정식이소개되었다. 예로써, Junger and Feit는 실린더의 벽이 매우 얇고 , 무한히 긴 실린더를 가정하에서 실린더운동방정식을 유동하였다. 본 논문에서는 참고문헌[5]에서 유도된 실린더운동방정식고 Junger and Feit 의 실린더운동방정식에서 실린더 좌표 방향으로 향하는 변위의 항을 비교하고, 임의의 실린더 크기를 가정하여 항의 차이를 그래프로 보였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.24-30
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• 2001

This paper describes a fast harmonic estimation, referred to as Delta Adjustment (DA), using a low resolution pitch. The presented DA method is based on modification of the Generalized Dual Excitation (GDE) technique[1] which was proposed to improve speech enhancement performance. We introduce the GDE technique and modify it to be suitable for low bit rate harmonic coding that uses only an integer pitch estimate. Unlike the GDE, the DA matches a frequency-warped version of the original spectrum that conforms to a fixed pitch at all harmonic bands. In addition, complexity and performance of the presented method are described in comparison with those of the conventional Fractional Pitch (FP) based harmonic estimation. Experimental results showed that the DA algorithm significantly reduces the complexity of the FP method while maintaining the performance.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.359-362
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• 1999

본 연구에서는 작은 삽입손실을 구현하기 위해 적용되는 SPUDT 중에서 현재 가장 널리 사용되고 좋은 성능을 가지는 DART와 EWC/SPUDT형 SAW 필터에 대해서 그 효용성을 Smith 등가회로 해석법을 이용하여 분석하였다. 그리고 apodization weighting된 SAW 필터에 단순히 SPUOT를 적용하였을 때 발생하는 일부 특성열화를 보상하기 위한 방법으로, 이 두 종류 SPUDT의 폭이 넓은 전극에 대한 weighting을 독립적으로 행하였고 전극두께를 조절하여 성능변화를 보았다. 그 결과로 동일 전극길이를 갖는 경우에는 EWC/SPUDT형 SAW 필터가 DART형 SAW 필터보다 삽입손실이 더 작으나 리플이 크고, apodization weighting된 SAW필터에 SPUDT를 적용한 경우에는 전반적으로 DART형 SAW 필터가 EWC/SPUDT형 SAW 필터보다 우수한 성능을 가지는 것으로 나타났다. 본 연구의 결과는 특정 형태의 SAW 필터에 SPUDT를 적용하는 문제에 있어서 적절한 SPUDT의 타입을 결정하는데 도움을 주리라 기대된다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.48-51
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• 1999

본 연구에서는 PCS 이동전화의 객관적 통화품질평가 척도개발을 위한 기초연구로 기존의 CD(Cepstral Distance), MSD (Mel Spectral Distance), BSD(Bark Spectral Distance), PSQM (Perceptual Speech Quality Measure) 척도를 적용하여 그 성능을 비교 분석하였다. 이 척도들을 실제환경에서 수집된 PCS 음성데이터에 대해서 적용하였고 이 결과치와 청취자들의 평가 반응에 의해 얻어진 MOS 결과치와의 상관성을 조사하였다. 실험 결과, BSD와 PSQM 척도의 상관성이 0.81, 0.84로 나타나 CD, MSD보다 성능이 더 우수함을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.20-28
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• 1995

This paper studied adaptive blind equalizer which belong to Bussgang type. It is well known that blind equalizer performs equalization without using a training sequence. Especially, this paper concentrated on real time processing of them. The channel characteristic was obtained from measurements taken in a real urban multipath environment. A T/2 fractionally-spaced equalizer was used at the receiving end. Our computer simulations demonstrated that Stop and Go, Benveniste-Goursat, and optimal Bussgang algorithms have relatively low MSE property. CMA shows faster convergence property than any other of Bussgang type algorithm.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.225-230
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• 1998

In this paper, a Kalman filter approach for enhancing speech signals degraded by statistically independent additive nonstationary noise is developed. The autoregressive hidden markov model is used for modeling the statistical characteristics of both the clean speech signal and the nonstationary noise process. In this case, the speech enhancement comprises a weighted sum of conditional mean estimators for the composite states of the models for the speech and noise, where the weights equal to the posterior probabilities of the composite states, given the noisy speech. The conditional mean estimators use a smoothing spproach based on two Kalmean filters with Markovian switching coefficients, where one of the filters propagates in the forward-time direction with one frame. The proposed method is tested against the noisy speech signals degraded by Gaussian colored noise or nonstationary noise at various input signal-to-noise ratios. An app개ximate improvement of 4.7-5.2 dB is SNR is achieved at input SNR 10 and 15 dB. Also, in a comparison of conventional and the proposed methods, an improvement of the about 0.3 dB in SNR is obtained with our proposed method.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.72-78
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• 2009

Heart sound segmentation into its components, S1, systole, S2 and diastole is the first step of analysis and the most important part in the automatic diagnosis of heart sounds. Conventionally, the Shannon energy envelope peak detection method has been popularly used due to its superior performance in locating S1 and S2. Recently, the HMM has been shown to be quite suitable in modeling the heart sound signal and its use in segmenting the heart sound signal has been suggested with some success. In this paper, we compared the two methods for heart sound segmentation using a common database. Experimental tests carried out on the 4 different types of heart sound signals showed that the segmentation accuracy relative to the manual segmentation was 97.4% in the HMM based method which was larger than 91.5% in the peak detection method.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.125-128
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• 2001

본 연구에서는 음성신호의 왜곡에 대해 음성 부재 확률을 고려한 MMSE(Minimum Mean Square Error) STSA(Short-Time Spectral Amplitude Estimator)를 전처리기로 도입하여 HMM(Hidden Markov Model)에 기반 한 음성인식시스템의 인식성능을 평가하였다. 음성인식 시스템의 실시간 구현을 고려하여, MMSE STSA 기법을 음성개선을 위한 전처리기로 사용할 때 MMSE STSA의 이득계산 과정에서 많은 계산량이 요구되는 modified Bessel 함수를 근사 화하여 사용하였다.

• The Journal of the Acoustical Society of Korea
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• v.15 no.3E
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• pp.37-44
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• 1996

A new method for efficient modeling of the Head-Related Transfer Functions(HRTF's) without loss of any directional information is proposed. In this paper, the HRTF's were empirically measured in a real room and modeled as the ARMA models with common AR coefficients and different MA coefficients. To assess the validity of the proposed ARMA model, psychophysical tests show that the proposed ARMA model, in comparison with the conventional MA model, requires a small number of parameters to represent empirical HRTF's and improves the back-to-front confusions in sound-field localization. Thus, significant simplifications in the implementations of sound-field synthesis systems could be obtained by using the proposed ARMA model.