• The Journal of the Acoustical Society of Korea
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• v.35 no.1
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• pp.55-62
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• 2016

The least-mean fourth (LMF) algorithm is well known for its fast convergence and low steady-state error especially in non-Gaussian noise environments. Recently, there has been increasing interest in the least mean square (LMS) algorithms with variable step size. It is because the variable step-size LMS algorithms have shown to outperform the conventional fixed step-size LMS in the various situations. In this paper, a variable step-size LMF algorithm is proposed, which adopts an algebraic optimal step size as a variable step size. It is expected that the proposed algorithm also outperforms the conventional fixed step-size LMF. The superiority of the proposed algorithm is confirmed by the simulations in the time invariant and time variant channels.

• The Journal of the Acoustical Society of Korea
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• v.41 no.4
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• pp.410-418
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• 2022

Channel reliability estimation for iterative codes such as turbo codes is very important factor in time varying underwater acoustic channel, an incorrect estimation of channel reliability induced performance degradation. Therefore, this paper presents an optimal channel reliability estimation method for turbo coded FSK signal with rate of 1/3. The estimated BER algorithm is a method that can estimate the reliability of received data by comparing received data and decoded data, and we determine optimal channel reliability by using the method. In order to analyze the performance, the experiment was conducted on a lake in Munkyeong city by moving in the range of 300 m to 500 m. At the result of applying presented method to failed decoding packets, we confirm all packets are decoded successfully.

• The Journal of the Acoustical Society of Korea
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• v.37 no.3
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• pp.156-162
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• 2018

The LMF (Least Mean Fourth) algorithm is well known for its fast convergence and low steady-state error especially in non-Gaussian noise environments. Recently, there has been increasing interest in the LMS (Least Mean Square) algorithms with self-adjusted step size. It is because the self-adjusted step-size LMS algorithms have shown to outperform the conventional fixed step-size LMS in the various situations. In this paper, a self-adjusted step-size LMF algorithm is proposed, which adopts an averaged gradient based step size as a self-adjusted step size. It is expected that the proposed algorithm also outperforms the conventional fixed step-size LMF. The superiority of the proposed algorithm is confirmed by the simulations in the time invariant and time variant channels.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.283-289
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• 2019

RLS (Recursive-least-squares) algorithm is known to have good convergence and excellent error level after convergence. However, there is a disadvantage that numerical instability is included in the algorithm due to inverse matrix calculation. In this paper, we propose an algorithm with no matrix inversion to avoid the instability aforementioned. The proposed algorithm still keeps the same convergence performance. In the proposed algorithm, we adopt an averaged gradient-based step size as a self-adjusted step size. In addition, a variable forgetting factor is introduced to provide superior performance for time-varying channel estimation. Through simulations, we compare performance with conventional RLS and show its equivalency. It also shows the merit of the variable forgetting factor in time-varying channels.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1
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• pp.86-97
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• 1998

본 논문은 스테레오 음향 반향 제거기에 적용되는 고속 Affine Projection 알고리듬 을 제안한다. 최근 스테레오 원격 회의 시스템은 보다 현실감 있는 원격 회의를 가능케 하 는 장점으로 인해 많은 관심을 끌고 있다. 그러나, 회의실의 원단화자와 마이크로폰사이의 상호교차(cross-coupling)로 인해 음향 반향이 발생하게 된다. 만약 이 반향 신호가 제거되 지 않은채 수신 룸으로 전달되면 결국 음성 통화 품질이 저하된다. 이를 방지하기 위하여 추정 반향 신호를 만들어 내고 통신 품질의 손실 없이 이 반향을 제거하는 음향 반향 제거 기가 필수적이다. 단 채널 음향 반향 제거기와 다르게 스테레오 환경하에서의 음향 반향 제 거기는 전송실의 환경변화로 인한 성능 저하와 각 반향 경로를 추정하기 위해 사용하는 각 적응 필터의 임펄스응답이 반향 경로와 일치하지 않는 등의 각종 문제점들이 발생하게 된 다. 본 논문에서는 서로 상관관계 없는 입력신호를 만들어내고 전송실의 환경변화로 인한 성능저하를 보완하기 위해 전처리단(pre-processing block)을 제안하여 일반적인 방법에 대 해 3-10dB정도의 향상된 성능을 보이며 적은 계산량으로 빠른 수렴성능을 갖는 새로운 형 태의 스테레오 음향 반향 제거기를 제안한다.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.04a
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• pp.141-144
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• 2000

스테레오 음향 반향 제거기는 채널신호간에 상호상관(cross-correlation) 때문에 적응필터 계수가 수신실의 반향 경로를 정확하게 추정하지 못한다. 본 논문에서는 상호상관을 줄이기 위해서 채널신호간 차의 절대값과 시변 감쇠상수를 이용하는 새로운 전처리 필터를 제안한다. 시뮬레이션을 통해서 제안하는 전처리 필터가 기존의 방법에 비해서 우수함을 보인다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.264-273
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• 2002

Adaptive filters cannot exactly estimate the echo path of the receiving room because of the cross-correlation of stereo signals. In this paper, a new pre-processing method reducing the cross-correlation without degradation of stereophony is proposed to enhance the performance of stereophonic acoustic echo canceller. To reduce the cross-correlation, absolutes of two orthogonal signals derived from each channel signals are added to original channel signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, the computation of pre-processing can be reduced. As results of simulations, it is shown that the performance of stereo acoustic echo canceller with the proposed pre-processing method is better than that of conventional ones.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.119-120
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• 2004

Split-beam 트랜스듀서의 입력신호로써 Sweep 신호를 구동한 어군탐지기 SONAR에 대하여 기술한다. Sweep 신호의 대역폭은 트랜스듀서 공진 대역의 대역폭에서 주파수 이동 변환하였다. 목표 대상물의 방위각 추정은 각 수신 채널 사이의 크로스-스펙트럼을 사용하여 추정한다. 추정 방위각의 정확도는 제안한 시스템의 알고리듬에 관계하고 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.377-380
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• 2004

정합장처리는 음파를 이용하여 해양환경 변수와 표적의 위치를 추정하는 신호처리 기법을 말한다. 이를 연구 하기 위하여 2003년 10월, 2004년 5월과 10월 등 3차례에 걸쳐 음향실험을 실시하였다. 동해시 앞의 수심 100-150 해당하는 대륙붕 지역에 20채널의 수직 선배열 센서를 설치하고 저주파 예인 음원기, 토모그래피 음원, 전구 발파기를 이용하여 음향신호를 획득하였다. 측정된 해양 및 음향자료와 실험 해역에 대한 지음향 데이터 베이스를 이용하여 일차적으로 음원의 위치 및 지음향 변수를 추정하는 분석을 실시하였다. 국방과학연구소, 한양대학교, 서울대학교가 참여하는 공동실험은 앞으로 3차례 더 실시할 계획에 있다.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.229-236
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• 2010

Phase coherent modulation techniques in underwater acoustic channel can improve bandwidth efficiency and data reliability, but they are made difficult by time-varying intersymbol interference. This paper proposes an adaptive equalizer combined with phase estimator which compensates distortions caused by time-varying multipath and phase variation. The experiment in the East sea demonstrates phase coherent signals are distorted by time-varying multipath propagation and the proposed scheme equalizes them. Bit error rate of BPSK and QPSK are 0.0078 and 0.0376 at 300 meter horizontal distance and 0.0146 and 0.0293 at 1000 meter respectively.