• Proceedings of the Acoustical Society of Korea Conference
• /
• autumn
• /
• pp.361-364
• /
• 2001

지금까지 수중음향 시스템에서 jammer 신호를 제거하는 방법에 관한 많은 연구가 진행되어 왔다. 그러나, 기존의 빔형성 기법은 간섭 신호원(interference source)이 움직일 경우 그 성능이 현저히 떨어지는 문제점을 갖고 있다. 이러한 현상은 수중 음향 시스템이 간섭 신호원의 움직임에 대하여 즉각적으로 null의 위치를 변화시키지 못하기 때문에 발생하게 된다. 이를 해결하기 위해서는 시간에 따라 위치가 변하는 jammer 환경에 대하여 대응할 수 있는 새로운 알고리즘이 필요하게 된다. 이러한 단점을 보완하기 위해 본 논문에서는 가변 망각인자를 갖는 적응 빔형성 기법을 제안하고, 컴퓨터 모의실험을 통하여 제안된 알고리즘이 기존의 적응 빔형성 기법에 비하여 출력 SINR(signal to interference plus noise ratio)의 측면에서 성능 향상을 가짐을 보였다.

• Journal of Sensor Science and Technology
• /
• v.9 no.6
• /
• pp.416-423
• /
• 2000

A new demodulation technique which can be used for the fiber optic acoustic sensor system using Sagnac interferometer is described. The theoretical limitation in dynamic range of the quadrature demodulation technique can be removed by the proposed BPSK(Binary Phase Shift Keying) demodulation technique. Full demodulation of input acoustic signal is possible with just simple electronics by eliminating the necessity of the high frequency phase modulation. This technique is suitable for digital signal processing of fiber optic sensor systems and can be applicable for other interferometers.

• Journal of the Korean Society of Surveying, Geodesy, Photogrammetry and Cartography
• /
• v.9 no.1
• /
• pp.63-73
• /
• 1991

The data type of the echo sounding for the contouring of coastal chart is continuous profiles, and there are no data between the profiles. In this study, at first, the depths of the regular grid along the sounding line were interpolated by linear equation. After that the depths of the regular grid located between the profiles were interpolated by kriging. The semivariogram contributes to the weight of interpolation. After comparison with the conventional Moving Average and Kriging, it turns out that this algorithm shows merits in the field of the accuracy and computing time.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.7
• /
• pp.600-605
• /
• 2002

In this paper, a Negative Impedance Converter (NIC) circuit was employed for the electro-mechanical characteristic control of a thickness mode piezoelectric vibrator. Two circular plane piezoelectric vibrators were bonded together and the NIC circuit was connected to one of the vibrators. The theoretical and experimental analysis of the characteristics shown that the quality factor and the electro-acoustic efficiency of the vibrator with the NIC circuit could be improved by 20 times and 2.5 times, respectively.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.83-86
• /
• 2002

훈련과정과 인식과정에서의 주변환경 잡음과 채널 특성 등의 불일치는 음성인식 성능을 급격히 저하시킨다. 이러한 불일치를 보상하기 위해서 켑스트럼 영역에서의 다양한 전처리 방법이 시도되고 있으며 최근에는 stereo 데이터와 잡음 음성의 Gaussian Mixture Model (GMM)을 이용해 보상벡터를 구하는 SPLICE 방법이 좋은 결과를 보이고 있다(1). 기존의 SPLICE가 전체 발성에 대해서 음향학적인 정보만으로 Gaussian 모델을 구하는 반면 본 논문에서는 발성에 해당하는 음소정보를 고려하여 전체 음향 공간을 각 음소에 대해 나누어서 모델링하고 각 음소에 대한 Gaussian 모델과 그 음소에 해당하는 음성데이터만을 이용하여 음소별 보상벡터가 훈련되도록 하였다. 이 경우 보상벡터는 잡음이 각 음소에 미치는 영향을 보다 자세히 나타내게 된다. Aurora 2 데이터베이스를 이용한 실험결과, 제안된 방법이 기존의 SPLICE방법에 비해 성능향상을 보였다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2000.11a
• /
• pp.49-62
• /
• 2000

• The Journal of the Acoustical Society of Korea
• /
• v.34 no.5
• /
• pp.394-402
• /
• 2015

CMA (Constant Modulus Algorithm) is one of the well-known algorithms in blind acoustic channel equalization. Generally, CMA converges slowly and the speed of convergence is dependent on a step-size in the CMA procedure. Many researches have tried to speed up the convergence speed by applying a variable step-size to CMA, e.g. the orthogonal projection CMA and algebraic optimal step-size CMA. In this paper, we summarize these two algorithms, and we propose a new CMA with improved convergence performance. The improvement comes from the switching between the orthogonal projection CMA and algebraic optimal step-size CMA. In simulation results, we show the performance improvement in the time invariant channels as well as in time varying channel.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.5
• /
• pp.35-43
• /
• 1999

In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.1
• /
• pp.1-10
• /
• 2012

The linear and decision-feedback equalization can mitigate time-varying intersymbol interference (ISI) caused by time-varying multipath propagation for underwater acoustic channels. The perfect elimination of interference components, however, is difficult using the linear equalization and the decision feedback equalizer has an error propagation problem. To overcome these shortcomings, this paper proposes an equalizer mode selection method using training sequences. The proposed method selects an equalization mode corresponding to the signal-to-noise ratio (SNR). If the SNR is low, the proposed system operates the linear equalizer for preventing the error propagation and if the SNR is high, the decision feedback equalizer for eliminating the residual ISI. Therefore, the proposed method can improve the error performance compared to the conventional equalizers. The computer simulation shows the proposed method improves the bit error performance using practical underwater channels responses acquired from the sea experiment.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.3A
• /
• pp.198-204
• /
• 2011

Acoustic communication system is a transmission technology sending sound and data simultaneously. However, data signal can be audible in this system when data is transmitted with high transmission power. The more transmission power is reduced, the more distance that can transmit data is shortened. Therefore, the study that increase the transmission distance is needed. In this paper, we would like to increase transmission distance by adapting receive diversity in acoustic communication system. We measure received performance of both proposed system and Single Input Sing Output (SISO) system according to distance with same transmission power. When SISO satisfies Bit Error Rate (BER) of $7{\times}10^{-3}$ at about 2m, Selection Combining (SC) technique satisfies 2 meters, and Equal Gain Combining (EGC) technique satisfies 4 meters.