최소 자승 평균오차와 부분 적응을 사용한 배열 빔 형성기법

임준석,최낙진,성굉모,김현석,Lim Jun-Seok,Choi Nakjin,Sung Koeng-Mo,Kim Hyun-Seok 한국음향학회 2004 한국음향학회지 Vol.23 No.8

In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.

기울기 벡터의 직교 정사형을 사용한 CMA 등화기에 관한 연구

임준석,Lim, Jun-Seok 한국음향학회 2009 韓國音響學會誌 Vol.28 No.7

블라인드 채널 등화기의 일종인 CMA방식의 수렴 속도는 일반적을 훈련열을 바탕으로 한 방법에 비해서 느린 경향이 있다. 또 그 수렴 속도는 정해진 스텝사이즈에 따라 정해진다. 본 논문은 기존 고정 스텝사이즈 CMA 알고리즘의 수렴 속도를 개선하기 위한 직교 정사형 망각인자를 도입하여 수렴성능이 개선된 새로운 알고리즘을 제안한다. CMA (Constant Modulus Algorithm) is one of the famous algorithms in blind 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. In this paper, we propose a new CMA algorithm with improved convergence performance. The improvement comes from an orthogonal projection of an average error gradient. We show the improvement in simulation results.

A Channel Equalization Algorithm Using Neural Network Based Data Least Squares

임준석,편용국,Lim, Jun-Seok,Pyeon, Yong-Kuk The Acoustical Society of Korea 2007 한국음향학회지 Vol.26 No.e2

Using the neural network model for oriented principal component analysis (OPCA), we propose a solution to the data least squares (DLS) problem, in which the error is assumed to lie in the data matrix only. In this paper, we applied this neural network model to channel equalization. Simulations show that the neural network based DLS outperforms ordinary least squares in channel equalization problems.

주파수영역 자기상관에 의한 위상 변환 일반 상호 상관 시간 지연 추정기 성능 개선

임준석,정명준,김성일,Lim, Jun-Seok,Cheong, MyoungJun,Kim, Seongil 한국음향학회 2018 한국음향학회지 Vol.37 No.5

There are several methods for estimating the time delay between incoming signals to two sensors. Among them, the GCC-PHAT (Generalized Cross Correlation-Phase Transform) method, which estimates the relative delay from the signal whitening and the cross-correlation between the different signal inputs to the two sensors, is a traditionally well known method for achieving stable performance. In this paper, we have identified a part of GCC-PHAT that can improve the periodicity. Also, we apply the auto-correlation method that is widely used as a method to improve the periodicity. Comparing the proposed method with the GCC-PHAT method, we show that the proposed method improves the mean square error performance by 5 dB ~ 15 dB at the SNR above 0 dB for white Gaussian signal source and also show that the method improves the mean square error performance up to 15 dB at the SNR above 2 dB for the color signal source.

잡음이 있는 두 음향 센서를 이용한 시간 지연 추정을 위한 향상된 적응 고유벡터 추정 기반 알고리즘

임준석,Lim, Jun-Seok 한국음향학회 2018 한국음향학회지 Vol.37 No.6

서로 떨어져 설치된 두 개의 음향 센서에 도달하는 신호의 상호 지연 시간을 추정하는 것은 실내 음향과 소나 등에서 목표물 위치 추정 문제나 추적 및 동기화에 이르기까지 다방면에서 쓰이고 있다. 시간 지연을 구하는 방법에서는 두 수신 신호 사이의 상호 상관을 이용한 방법이 대표적이다. 그러나 이 방법은 수신 음향 센서에 잡음이 부과 되는 것에 충분한 고려가 없었다. 본 논문은 수신 음향 센서에 모두 잡음이 부과된 경우를 고려한 새로운 시간 지연 추정 방법을 제안한다. 기존의 일반 상호 상관법과 적응 고유치 분석법과 비교를 통해서 새로 제안한 알고리즘이 유색 신호에 부가된 가우시안 잡음환경에서 우수성이 있음을 확인한다.

희소성 음향 통신 채널 추정 견실화를 위한 백색화를 적용한 l₁놈-RLS 알고리즘

임준석,편용국,김성일,Lim, Jun-Seok,Pyeon, Yong-Gook,Kim, Sungil 한국음향학회 2020 한국음향학회지 Vol.39 No.1

This paper proposes a new l₁-norm-Recursive Least Squares (RLS) algorithm which is numerically more robust than the conventional l₁-norm-RLS. The l₁-norm-RLS was proposed by Eksioglu and Tanc in order to estimate the sparse acoustic channel. However the algorithm has numerical instability in the inverse matrix calculation. In this paper, we propose a new algorithm which is robust against the numerical instability. We show that the proposed method improves stability under several numerically erroneous situations.

비선형 전처리와 주파수 영역 저역 필터에 의한 임펄스성 잡음 환경에 강인한 위상 변환 일반 상호 상관 시간 지연 추정기 연구

임준석,이근화,Jun-Seok Lim,Keunwa Lee 한국음향학회 2024 韓國音響學會誌 Vol.43 No.4

본 논문은 임펄스성 잡음에 견실한 시간 지연을 추정 방법을 제안한다. 제안하는 방법은 Generalized Cross Correlation - Phase Transform(GCC-PHAT) 방법에 비선형 전처리와 주파수 영역 저역 통과 필터를 사용한다. 본 논문에서는 GCC-PHAT의 연산 과정을 DFT로 다시 해석함으로써 GCC-PHAT에서 시간 지연 추정에 실제 사용되는 유효한 주파수 대역이 있음을 파악하고, 저역 통과 필터 사용하여 유효 대역만을 사용하면 잡음 성분을 줄여서 시간 추정 성능을 향상시킬 수 있음을 보인다. 제안한 방법을 임펄스성 잡음 환경에서 전통적인 GCC-PHAT와 비교함으로써 추정 견실성이 향상됨을 보인다. The proposed method uses Generalized Cross Correlation - Phase Transform (GCC-PHAT) method with nonlinear preprocessing and a frequency domain low-pass filter. In this paper, by reinterpreting the calculation process of GCC-PHAT as DFT, we derive that there is an effective frequency band used for time delay estimation in GCC-PHAT, and by using only the effective band using a low-pass filter, the noise component is reduced and it improvesthe time delay performance in impulsive noise environments. By comparing the proposed method with the traditional GCC-PHAT in an impulsive noise environment, we show that the GCC-PHAT becomes more robust to the impulsive noise.

RLS (Recursive Least Squares)와 RTLS (Recursive Total Least Squares)의 결합을 이용한 새로운 FIR 시스템 인식 방법

임준석,편용국,Lim, Jun-Seok,Pyeon, Yong-Gook 한국음향학회 2010 韓國音響學會誌 Vol.29 No.6

잡음이 섞인 입출력 신호를 갖는 시스템 인식 문제는 완전 최소 자승법 (Total Least Squares (TLS))으로 알려져 있다. 완전 최소 자승법의 성능은 입력 신호 부가 잡음 파워와 출력 신호 부가 잡음간의 분산비에 매우 민감하다. 본 논문에서는 TLS의 성능 향상을 위해서 LS (Least Squares)와의 결합을 제안한다. 그 한 형태로 재차적인 TLS (Recursive TLS)와 재차적인 LS (Recursive Least Squares)간의 결합 알고리즘을 제안한다. 이 결합은 잡음간 분산비에 강인한 결과를 낳았다. 모의실험을 통해 얻은 결과로부터 입력 신호에 신호대 잡음비가 5dB를 유지히는 잡음을 부가할 경우 입력 잡음과출력 잡음의 비 $\gamma$가 약 20 정도까지로 적용 범위가 확대되는 결과를 얻었다. 따라서 제안된 결합 방법이 기존의 TLS의 적용 범위를 넓힐 수 있음을 알 수 있다. It is known that the problem of FIR filtering with noisy input and output data can be solved by a total least squares (TLS) estimation. It is also known that the performance of the TLS estimation is very sensitive to the ratio between the variances of the input and output noises. In this paper, we propose a convex combination algorithm between the ordinary recursive LS based TLS (RTLS) and the ordinary recursive LS (RLS). This combined algorithm is robust to the noise variance ratio and has almost the same complexity as the RTLS. Simulation results show that the proposed algorithm performs near TLS in noise variance ratio ${\gamma}{\approx}1$ and that it outperforms TLS and LS in the rage of 2 < $\gamma$ < 20. Consequently, the practical workability of the TLS method applied to noisy data has been significantly broadened.

반복형 위너 필터 방법에 기반한 재귀적 완전 최소 제곱 방법을 사용한 시간 지연 추정 알고리즘

임준석,Lim, Jun-Seok 한국음향학회 2021 한국음향학회지 Vol.40 No.5

Estimating the mutual time delay between two acoustic sensors is used in various fields such as tracking and estimating the location of a target in room acoustics and sonar. In the time delay estimation methods, there are a non-parametric method, such as Generalized Cross Correlation (GCC), and a parametric method based on system identification. In this paper, we propose a time delay estimation method based on the parametric method. In particular, we propose a method that considers the noise in each receiving acoustic sensor. Simulation confirms that the proposed algorithm is superior to the existing generalized cross-correlation and adaptive eigenvalue analysis methods in white noise and reverberation environments.

Basis pursuit denoising을 사용한 두 수신기 간 시간 지연 추정 알고리즘

임준석,정명준,Lim, Jun-Seok,Cheong, MyoungJun 한국음향학회 2017 한국음향학회지 Vol.36 No.4

Many methods have been studied to estimate the time delay between incoming signals to two receivers. In the case of the method based on the channel estimation technique, the relative delay between the input signals of the two receivers is estimated as an impulse response of the channel between the two signals. In this case, the characteristic of the channel has sparsity. Most of the existing methods do not take advantage of the channel sparseness. In this paper, we propose a time delay estimation method using BPD (Basis Pursuit Denoising) optimization technique, which is one of the sparse signal optimization methods, in order to utilize the channel sparseness. Compared with the existing GCC (Generalized Cross Correlation) method, adaptive eigen decomposition method and RZA-LMS (Reweighted Zero-Attracting Least Mean Square), the proposed method shows that it can mitigate the threshold phenomenon even under a white Gaussian source, a colored signal source and oceanic mammal sound source.