• Transactions of the Korean Society of Mechanical Engineers
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• v.15 no.3
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• pp.809-823
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• 1991

본 연구에서는 Wiener 필터링 이론에 의하여 소음원의 입력신호에 대한 최적 한 예측을 할 수 있는 최적예측기(optimal predictor)로써 부가적인 음을 발생시키고 입력신호 및 출력신호 간의 차인 오차를 최소화시키도록 하는 적응신호처리방법에 대 해 설명하고 이러한 적응 신호처리 방법을 이용한 능동 소음 제어 방법을 제시하였다. 이와 아울러 제어계의 환경 변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해 야 하는 적응 소음 제어계에서, 음향궤환과 함께 필히 고려해야하는 부가적인 전달함 수-모델과 스피커를 포함하는 보조경로 및 오차미이크로폰을 포함하는 오차경로의 전 달함수의 영향을 고려한 능동소음제어에 대해 연구하였다.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.42 no.4 s.304
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• pp.45-50
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• 2005

In digital hearing aid system, one of the main problems is acoustic feedback which is known as howling because of miniaturization md high-gain amplification. In this paper, we proposed a feedback cancellation algorithm for hearing aid using time delay and time-varying decorrelation filter. The proposed algorithm has a kind of adaptive filter structure, which is combined with time delay and time-varying decorrelation filter to improve feedback cancellation. An all pass filter was implemented as the time-varying decorrelation filter using low frequency modulator. From the result of computer simulation, it is verified that the proposed algorithm has good ability to cancel feedback.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.10a
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• pp.565-568
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• 2008

In this paper, we present bit error rate(BER) performance of an adaptive decision feedback equalizer(DFE) with experimental data. The experiment was performed at the shore of Geoje in November 2007. The BER performance of the adaptive DFE whose tap weight is updated by RLS is described with change of feedforward tap number, feedback tap number, traning seqence length and delay, which shows that the uncoded average BER is $4{\times}10^{-2}\;and\;1.5{\times}10^{-2}$ with transmission range 9.7km and 4km, respectively.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.699-702
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• 2009

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows the improved SNR of about more than 20 dB.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.4
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• pp.645-651
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• 2009

In this paper, we present bit error rate(BER) performance of an adaptive decision feedback equalizer(DFE) using experimental data. The experiment was performed at the shore of Geoje in November 2007. The BER of the adaptive DFE whose tap weight is updated by RLS is described with change of feedforward filter length, feedback filter length, training sequence length, and delay, which shows that the uncoded average BER is $4{\times}10^2\;and\;1.5{\times}10^{-2}$ with transmission range of 9.7km and 4km, respectively. The BER of the adaptive DFE can be lower than 10-3 by a forward error correction code and therefore the adaptive DFE may be a good candidate for a high speed AUV communications since the volume and weight of the underwater acoustic modem should be small because of the restricted space and power in the battery-operated AUV.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1994.10a
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• pp.331-336
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• 1994

본 논문에서는 음향궤환 문제해결을 위한 한 방법으로써, 넓은 주파수 영역에서 단일지향특성을 가질 수 있도록 주파수영역에서 음파분리를 하여, 하류측으로 전파되는 음파를 제어하는 새로운 방법을 제시한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.8
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• pp.1765-1772
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• 2015

In the underwater acoustic communication channels, multipath reflection become the cause of obstacle. Generally, equalizer has been applied to overcome these problems. In this paper, the method was proposed to optimize tap-length of decision feedback equalizer using genetic algorithm. After inputting feed-forward filter length and feed-back filter length as genetic information of the genetic algorithm, it optimize tap-length using BER(bit error rate) calculation in accordance with object function. The object function consist of decision feedback equalizer and BER calculation. For the purpose of BER calculation in the object function, the method was proposed to optimize the tap-length of decision feedback equalizer with genetic algorithm using preamble signals. As a result of experiments, the optimized BER is 0.0355 for signals which were received through a 25m receiver and which were applied to calculate BER merely using preamble signals in object function. When all data were used to calculate BER in object function, the optimized BER is 0.0215.

• The Journal of the Acoustical Society of Korea
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• v.27 no.8
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• pp.443-449
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• 2008

One of the key issues in practical speech processing is to achieve robust Voice Activity Deteciton (VAD) against the background noise. Most of the statistical model-based approaches have tried to employ equally weighted likelihood ratios (LRs), which, however, deviates from the real observation. Furthermore voice activities in the adjacent frames have strong correlation. In other words, the current frame is highly correlated with previous frame. In this paper, we propose the effective VAD approach based on a minimum classification error (MCE) method which is different from the previous works in that different weights are assigned to both the likelihood ratio on the current frame and the decision statistics of the previous frame.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.110-116
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• 1996

In this paper a novel method improving the accuracy of DTMF tone reception in ARS system is proposed. ARS system should allow users to generate DTMF signals while it is sending voice guidance. It is not unocmmon, in this case, that a portion of transmitting voice signals cross-talks to the receiving channel and it often results in interfering with the receiving DTMF signals. Serious degradations including DTMF tone missing, false alarm and so forth have been introduced for the above reason. To overcome this phenomena, we have proposed a way eliminating the frequency spectra representing DTMF signals bands from the transmitting voice signal by using notch filters. This method also employs bandpass filters of which the frequency responses are reciprocal to those of the notch filters incorporated with the DTMF receiver. It is shown that a drastic improvement has been achieved with respect to the DTMF tone detection with little deterioration of voice guidance quality.