• The Journal of the Acoustical Society of Korea
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• v.34 no.6
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• pp.479-486
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• 2015

A triplet line array has a capability for solving left/right ambiguity problem and high directivity of line array. For this, high computational powers and correlation between nearby sensors must be considered. Hardware powers are dramatically increased, so various adaptive beamforming techniques and signal processing techniques can be adopted for large triplet line array. To investigate the optimum signal processing method and improve the target detection capability, we need to analyze the noise correlation among the triplet line array sensors in real environment. In this paper we analyze the experimental data in the East Sea, investigate noise correlation between three hydrophones in triplet sensors, among the triplet sensors and time varying properties. Based on the acquired results, it is possible to verify some information for beamforming and signal processing methods with considering the properties of the triplet line array.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.5-10
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• 1997

In this paper, a new time delay estimation algorithm, WTD-LMSTDE was proposed. This method has great improvement in convergence rate relative to the time domain approach by decreasing the eigen value spread of input signal autocorrelation matrix. The performance of the algorithm was evaluated for the cases of time invariant time delay and time varying time delay. In the case of time invariant time delay, the estimation accuracy of WTD-LMSTDE was better than that of LMSTDE from 3.3% to 12.5% with respect to SNR. In the case of time varying time delay, the mean error power of WTD-LMSTDE in linear increased delay environment was decreased about 2.4dB compared to that of LMSTDE under noise-free condition. As a result, we showed that the performance of WTD-LMSTDE is better than of LMSTDE.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.48-53
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• 2014

The underwater acoustic communication (UAC) is characterized by doubly spread channel. It is included in the time-variant doppler shift and delay-time spreads due to multiple paths. To compensate such distorted signals, various techniques including time-reversal processing, spatial diversity, phase estimator, and equalizer are being applied. In this paper, a spatial filter based on the beamforming is proposed as a method to mitigate such inter-symbol interferences that are generated in time-varying multipath channels. The proposed technique realizes coherent communications by steering the direction of the desired signals and improves the performance of UAC by increasing the signal-to-interference plus noise ratio using the array gain.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5A
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• pp.618-633
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• 2000

In this paper, a Gaussian interpolation filter and cubic interpolation filter are presented to do more accurate channel estimation compared to the conventional linear interpolation filter for COFDM systems. In addition to an interpolation filter, a low pass filter using FFT and IFFT is also presented to reduce the noisy components of a channel estimate obtained by an interpolation filter. Channel estimates after low-pass filtering combined with interpolation filters can lower the error floor compared to the use of only interpolation filters. Computer simulation demonstrates that the presented channel estimation methods exhibit an improved performance compared to the conventional linear interpolation filter for COFDM systems in time-varying multipath fading channel and0.1 ~ 0.2 dB of Eb/No difference at BER=10-4 when the perfect channel estimation is compared.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.539-542
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• 2011

In this paper, we have implemented a performance analysis of $2{\times}1$ Alamouti scheme suggested by Alamouti, composed of the transmit space-time code and the simple linear decoding processing, in perfectly time-varying and spatially correlated channels. In addition, we derived the closed-form probability density function (PDF) of the output signal-to-noise ratio (SNR) and the outage probability of the Alamouti scheme as a function of the spatial correlation coefficient in the consideration of no correlation in time. As a result, it was found that the performance of the Alamouti scheme could be significantly degraded particularly in the case that the channels are time-varying and spatially correlated.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.473-480
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• 2004

In this paper we propose an effective feature compensation scheme based on the speech model in order to achieve robust speech recognition. The proposed feature compensation method is based on parallel combined mixture model (PCMM). The previous PCMM works require a highly sophisticated procedure for estimation of the combined mixture model in order to reflect the time-varying noisy conditions at every utterance. The proposed schemes can cope with the time-varying background noise by employing the interpolation method of the multiple mixture models. We apply the‘data-driven’method to PCMM tot move reliable model combination and introduce a frame-synched version for estimation of environments posteriori. In order to reduce the computational complexity due to multiple models, we propose a technique for mixture sharing. The statistically similar Gaussian components are selected and the smoothed versions are generated for sharing. The performance is examined over Aurora 2.0 and speech corpus recorded while car-driving. The experimental results indicate that the proposed schemes are effective in realizing robust speech recognition and reducing the computational complexities under both simulated environments and real-life conditions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7A
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• pp.689-695
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• 2006

This paper focuses on the accurate performance evaluation of cooperative diversity technique with imperfect channel estimates. The channel environment for simulations and performance evaluation is supposed to be the slowly time-varying Rayleigh fading channel. The framework of the performance evaluation is based on the Moment Generating Function(MGF) approach. To apply the effect of this channel estimation error into the performance evaluation, we import an useful Gaussian approximation in formulating the effective noise component and the additive noise. The average BER performance of cooperative diversity with M-PSK and M-QAM is computed as a function of the ratio of the signal to the effective noise based on the approximation. The verification of computed performance is provided with simulations. The evaluated performance matches up to simulation results even in a low SNR region.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1210-1215
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• 2000

Investigation is performed on the stability of general form of dynamic system under colored noise random disturbance whose damping and stiffness are varying in irregular manner along time, which is a preliminary result in the course of research on the characteristic and the control of the stochastic system. Adopted physical model is airfoil under random atmospheric disturbance, which becomes a "time-varying system" whose the governing equation is derived via F-P-K approach in stochastic sense. Control performance and effect of 'Heo-stochastic controller for colored noise' is studied. Also stochastic feature of flutter boundary is discussed as well.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.163-166
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• 2001

초음파를 이용하여 인체 조직의 특성을 알아내는 방법은 매우 광범위하게 응용되어 오고있다. 그 중에서 초음파를 발생시킨 후 반사되어 되돌아오는 신호를 측정하여 그 감쇠 정도로부터 조직의 특성을 추정하는 방법이 많이 사용되고 있다. 이러한 감쇠현상은 발생된 초음파가 조직 내에서 흡수 또는 산란현상을 거치면서 주파수가 변이를 일으키기 때문에 발생한다. 따라서, 조직의 감쇠 특성을 알아내기 위해서, 주파수의 함수로 근사할 수 있는 감쇠 계수(attenuation coefficient)를 이용하여 시간에 따라 달라지는 주파수 변화를 추정한다. 그러나, 기존의 Ah(Auto-Regressive) 모델을 통한 시간영역 및 주파수 영역에서의 추정 방법을 사용하면 잡음이 존재하는 상황에서 시변 신호를 추정하는데 성능이 많이 저하된다. 본 논문에서는 이러한 단점을 보완하기 위해서, 가변 망각 인자와 재귀적인 TLS(Total Least Squares) 방법을 사용하여 시간에 따라 변하는 신호를 정확하게 추정하고 잡음환경에도 강인한 알고리듬을 제안하였다. 또한, 제안된 알고리듬은 추정 성능을 향상시킬 뿐 아니라 감쇠정도의 강약에 관계없이 망각인자의 값을 적응적으로 변화시켜 동작하는 장점을 가진다.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.4 no.2
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• pp.82-90
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• 1993

전자파에 의한 산란현상의 해석은 지금까지 주로 시간조화함수의 형태를 지닌 전원에 의한 정 상상태의 산란에 관하여 이루어졌다. 그러나 레이다나 피파괴 검사, 전송선로 점검 등의 응용에서는 주로 펄스형태의 전자파를 사용하며, 따라서 시간에 따라 변화하는 함수형태의 전원에 의한 전자파의 산란해 석이 중요한 문제로 등장하였다. 또한 통신선로에서 외부의 잡음에 대한 혼신 등을 해석하거나, 낙뢰가 송 전선로에 미치는 영향을 해석하는 데에도 펄스신호의 산란해석이 필수적이다. 일반적인 함수의 형태를 지닌 전원에 의한 산란현상을 해석하기 위해서는 전원함수를 Fourier 변환하 여 주파수 영역의 스펙트럼을 구하고, 주파수영역에서의 산란해를 이용하여 Fourier 역변환을 하여 시간 영역의 해를 구할 수 있다. 주파수 영역에서의 산란판의 해를 Fourier 역변환 하기 위해서는 적분을 행하여야 하며, 일반적으로 적분과정에서 매우 복잡한 계산이 필요하고, 산란체의 구조가 복잡하여 해석 적인 해를 구할수 없는 경우에는 해석적으로 시간영역의 해를 구하는 것이 불가능하다. 시변 함수에 의 한 산란파를 구하기 위한 수치해석적 방법으로는 모멘트방법이나 유한요소법(Finite Element Method), 경계요소법(Boundary Element Method), 유한차분법(Finite Difference Method)등이 있으며, 해석적 해 를구할 수 없는 경우에 적용할 수 있는 반면에 많은 계산량이 요구된다.