• Transactions of the Korean Society of Mechanical Engineers
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• v.14 no.2
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• pp.331-338
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• 1990

In adaptive control, the lack of persistent and rich excitation causes the estimated parameters to drift, which degrade the performance of the system and may introduces instability to the system in a stochastic environment. To solve the problem of the parameter drift, the concept of single parameter adaptation is presented. For the parameter identification, a priori error is directly used for adaptation error. The structure of the controller is based upon the minimum variance control technique. The stability and robustness analysis is carried out by the sector stability theorem for the second order system. The computer simulation is performed to justify the theoretical analysis for the various cases.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.15-22
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• 1992

잡음이 있는 환경하에서는 음성 인식의 성능이 현저하게 떨어지게 된다. 본 논문에서는 이렇나 잡음의 영향에 강한 거리척도를 제안하고자 한다. 우리는 잡음이 더해진 음성신호의 특징벡터를 깨끗한 음성신호의 특징벡터가 FIR 시스템을 거쳐 변형된 것이라고 가정한다. 여기서 FIR 시스템은 잡음의 영 향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템 계수잡음의 영향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템계수들은 RLS 적응 알고리즘을 이용하여 구한다. 제안된 거리척도는 적응 여파 기의 예측 오차에 관한 식으로 표시되어진다. 여러 가지 적응 여파기의 구조중 단일 채널 일차 FIR 구 조가 가장 좋은 음성 인식 성능을 보이며, 이 경우 효과적인 거리척도 알고리즘을 구할 수 있다. 여러 가지 신호대 잡음비에 관하여 화자독립 격리단어 인식 실험을 DTW 알고리즘을 이용하여 수행하여 본 결과 제안된 거리척도가 거의 모든 신호대 잡음비에 대하여 우수한 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.34 no.4
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• pp.310-315
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• 2015

A single channel VAD (Voice Activity Detection) algorithm for nonstationary noise environment is proposed in this paper. Threshold values of the feature parameter for VAD decision are updated adaptively based on estimates of means and standard deviations of past non-speech frames. The feature parameter, SPD-TE (Spectral Power Difference-Teager Energy), is obtained by applying the Teager energy to the WPD (Wavelet Packet Decomposition) coefficients. It was reported previously that the SPD-TE is robust to noise as a feature for VAD. Experimental results by using TIMIT speech and NOISEX-92 noise databases show that decision accuracy of the proposed algorithm is comparable to several typical VAD algorithms including standards for SNR values ranging from 10 to -10 dB.

• Journal of Broadcast Engineering
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• v.24 no.4
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• pp.613-622
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• 2019

A differential codebook design method using wireless channel's temporal correlation is proposed over closed loop multiple-input single-output (MISO) system. The single layer codewords in a codebook are selected among a set of phase elements. In the conventional codeword selection rule, codewords are assumed to be on a spherical cap and sine formula was used. In this paper, however, a new method using Pythagoras formula is employed to simplify computational complexity. Also, an adaptive differential codebook selection is adopted to enhance performance. Monte-Carlo simulations demonstrate that the proposed codebook is superior to the conventional ones.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.3
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• pp.71-78
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• 2009

In this study, estimation of heart rate from measured PPG signal is proposed. PPG signal is to be measured blood flow in a blood vessel effected by systole and diastole. PPG sianl has single frequency so that PPG frequency can be tracked by 2nd IIR adaptive notch filter. PPG frequency is obtained continually from updating filter coefficient throughout adaptive algorithm and then the heart rate of human is approximately estimated.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.45-54
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• 1999

본 논문에서는 웨이브렛 변환 대역에서 영역분할 기법을 적용하여 얻어진 각 영역을 중요 영역과 비 중요 영역으로 분류하고 각각의 영역을 그의 특성에 적합한 방식으로 부호화 하는 기법을 제안하였다. 중요 영역은 전체 영역가운데 매우 작은 부분을 차지하지만 영상 복원에 매우 큰 영향을 주기 때문에 이러한 영역 부호화를 위해 기존의 EZW 방식보다 성능이 우수하며 단일계수 전송에 성능이 뛰어난 SPIHT 알고리즘을 적용하였다. 그러나 비 중요영역은 영상복원에 미치는 영향이 적을 뿐만 아니라, 매우 큰 동질 영역을 형성하기 때문에 텍스춰 모델링을 이용할 경우 높은 압축률을 얻을 수 있다. 또한 이 방식을 이용할 경우 인위적인 에러가 거의 없기 때문에 이용할 경우 높은 압축률을 얻을 수 있다. 또한 이 방식을 이용할 경우 인위적인 에러가 거의 없기 때문에 시각적으로도 좋은 영상을 복원 할 수 있다. 실험결과 제안한 시스템은 다양한 영상에 대하여 적응성이 있음을 보였고 특히 0.2bpp 이하의 매우 낮은 비트 율에서도 EZW 와 같은 기존의 웨이브렛 기반 부호화기보다 좋은 성능을 나타내었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.5C
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• pp.478-486
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• 2007

Double-talk detection errors can result in either large residual echo or distorting the near-end talker's input speech. Thus accurate double-talk detection is an important problem in the acoustic echo canceller to improve the speech quality. In the double-talk detection algorithm using a cross-correlation coefficient, double-talk detection errors can occur in the initial convergence period of an adaptive filter or in noisy environment since the cross-correlation coefficient becomes large in such situations. In this paper, we propose a new double-talk detection algorithm based on the cross-correlation method using a normalized error signal power to reduce the double-talk detection errors. The experimental results have shown the performance improvement of an acoustic echo canceller as well as the noise-robustness of the proposed double-talk detector.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.57-63
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• 1997

This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1471-1479
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• 1999

This paper presents an adaptive beamforming algorithm for an MC-CDMA system with an adaptive array antenna. By employing an antenna array at the receiver of an MC-CDMA system, the performance of an MC-CDMA system, which is known to be effective for high data rate transmission due to its robustness to multipath fading and its simplicity for using a simple one-tap equalizer, is shown to be significantly improved. The proposed algorithm for adaptive beanforming in an MC-CDMA system is derived by (1) calculating the error signals between the pilot symbols of desired user and the received pilot signals in frequency domain, (2) transforming the frequency-domain error signals into time-domain error signals, (3) updating the filter coefficients of the adaptive beamformer in the direction of minimizing the MSE. Convergence behavior and performance improvement of the proposed approach are demonstrated through computer simulation by applying it to the conventional MC-CDMA system.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.3-11
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• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.