• Title/Summary/Keyword: array response estimation

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Spatial Spectrum Estimation of Incident Signal Via Measured Array Manifold (측정 Array Manifold를 적용한 입사 신호의 공간 스펙트럼 추정)

  • 강흥용;이성윤;김영수;김창주;박한규
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.15 no.3
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    • pp.223-230
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    • 2004
  • A method for measuring array manifold which is the array antenna response of incident signal is presented. Array manifold measurement procedure by the presented method is explained for UCA(Uniform Circular Array), and spatial spectrum of 300 ㎒ tone signal incident on UCA is estimated by MUSIC algorithm in which spatial spectrum peak is searched with measured array manifold. Spatial spectrum estimation using array manifold measured by the proposed method shows superior performance to calculated array manifold.

Direction of Arrival Estimation for Desired Target to Remove Interference and Noise using MUSIC Algorithm and Bayesian Method (베이즈 방법과 뮤직 알고리즘을 이용한 간섭과 잡음제거를 위한 원하는 목표물의 도래방향 추정)

  • Lee, Kwan-Hyeong;Kang, Kyoung-Sik
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.8 no.5
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    • pp.400-404
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    • 2015
  • In this paper, we study for direction of arrival MUSIC spatial spectrum algorithm in order to desired signal estimation in spatial. Proposal MUSIC spatial spectrum algorithm in paper use model error and Bayesian method to estimation on correct target position. Receiver array response vector using adaptive array antenna use Bayesian method, and target position estimate to update weight value with model error method. Target's signal estimation of desired direction of arrival in this paper apply weight value of signal covariance matrix for array response vector after removing incident signal interference and noise, respectively. Though simulation, we analyze to compare proposed method with general method.

Joint Estimation of Near-Field Source Parameters and Array Response

  • Cui, Han;Peng, Wenjuan
    • Journal of Information Processing Systems
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    • v.13 no.1
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    • pp.83-94
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    • 2017
  • Near-field source localization algorithms are very sensitive to sensor gain/phase response errors, and so it is important to calibrate the errors. We took into consideration the uniform linear array and are proposing a blind calibration algorithm that can estimate the directions-of-arrival and range parameters of incident signals and sensor gain/phase responses jointly, without the need for any reference source. They are estimated separately by using an iterative approach, but without the need for good initial guesses. The ambiguities in the estimations of 2-D electric angles and sensor gain/phase responses are also analyzed in this paper. We show that the ambiguities can be remedied by assuming that two sensor phase responses of the array have been previously calibrated. The behavior of the proposed method is illustrated through simulation experiments. The simulation results show that the convergent rate is fast and that the convergent precision is high.

Differential Code-Filtering Correlation Method for Adaptive Beamforming

  • Hefnawi Mostafa;Denidni Tayeb A.
    • Journal of Communications and Networks
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    • v.7 no.3
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    • pp.258-262
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    • 2005
  • An adaptive beamforming system based on code filtering and differential correlation approaches is proposed. The differential correlation method was originally proposed for time delay estimation of direct sequence code division multiple access (DS-CDMA) systems under near-far ratio conditions and the code filtering correlation algorithm, on the other hand, was proposed for array response estimation in DS-CDMA systems under perfect power control. In this paper, by combining differential correlation concept with the code filtering beamforming technology, an accurate estimate of the beam forming weights and an enhanced performance of DS-CDMA systems under sever near-far ratio conditions is achieved. The system performance in terms of beam pattern and bit-error-rate (HER) shows that the proposed adaptive beamformer outperforms the conventional code filtering correlation technique.

A Study on Signal Sub Spatial Method for Removing Noise and Interference of Mobile Target (이동 물체의 잡음과 간섭제거를 위한 신호 부 공간기법에 대한 연구)

  • Lee, Min-Soo
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.8 no.3
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    • pp.224-228
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    • 2015
  • In this paper, we study the method for desired signals estimation that array antennas are received signals. We apply sub spatial method of direction of arrival algorithm and adaptive array antennas in order to remove interference and noise signal of received antenna signals. Array response vector of adaptive array antenna is probability, it is correctly estimation of direction of arrival of targets to update weight signal. Desired signals are estimated updating covariance matrix after moving interference and noise signals among received signals. We estimate signals using eigen decomposition and eigen value, high resolution direction of arrival estimation algorithm is devided signal sub spatial and noise sub spatial. Though simulation, we analyze to compare proposed method with general method.

A Calibration Technique for Array antenna based GPS Receivers (배열 안테나 기반 GPS 수신기에서의 교정 방안)

  • Kil, Haeng-bok;Joo, Hyun;Lee, Chulho;Jeong, Eui-Rim
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.22 no.4
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    • pp.683-690
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    • 2018
  • In this paper, a new signal processing technique is proposed for calibrating gain, phase, delay offsets in array antenna based anti-jamming minimum variance distortionless response (MVDR) global-positioning-system (GPS) receivers. The proposed technique estimates gain, phase and delay offsets across the antennas, and compensates for the offsets based on the estimates. A pilot signal with good correlation characteristics is used for accurate estimation of the gain, phase and delay offsets. Based on the cross-correlation, the delay offset is first estimated and then gain/phase offsets are estimated. For fine delay offset estimation and compensation, an interpolation technique is used, and specifically, the discrete Fourier transform (DFT) is employed for the interpolation technique to reduce the computational complexity. The proposed technique is verified through computer simulation using MATLAB. According to the simulation results, the proposed technique can reduce the gain, phaes and delay offset to 0.01 dB, 0.05 degree, and 0.5 ns, respectively.

Estimation of Wave Energy Extraction Efficiency for a Compact Array System of Small Buoys (밀집 배열 부이시스템의 파랑에너지 추출 효율 추정)

  • Choi, Yoon-Rak
    • Journal of Ocean Engineering and Technology
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    • v.25 no.1
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    • pp.8-13
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    • 2011
  • A compact array system of small buoys is used for wave energy extraction. To evaluate the performance of this system, hydrodynamic analysis is carried out in regular waves using the higher order boundary element method. The motion response of each buoy is calculated considering hydrodynamic interactions caused by other buoys. The effect of energy extraction device is modeled as a linear damping load. The efficiencies of energy conversion are compared using the various sizes and arrangements of the array system and the damping coefficients for energy extraction. The increase in size or the packing ratio of the system gives better efficiency. However, the wave condition and the cost for the system should be considered to optimize performance from the perspective of engineering and economics. The proposed nondimensionalized damping coefficient for energy extraction is 0.1~0.5.

An online Calibration Algorithm using binary spreading code for the CDMA-based Adaptive Antenna Array

  • Lee, Chong-Hyun
    • Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.20 no.9
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    • pp.32-39
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    • 2006
  • In this paper, an iterative subspace-based calibration algorithm for a CDMA-based antenna array in the presence of unknown gain and phase error is presented. The algorithm does not depend on the array geometry and does not require a prior knowledge of the Directions Of Arrival (DOA) of the signals. The method requires the code sequence of a reference user only. The proposed algorithm is based on the subspace method and root finding approach, and it provides estimates of the calibration vector, the DOA and the channel impulse response, by using the code sequence of a reference user. The performance of the proposed algorithm was investigated by means of computer simulations and was verified using field data measured through a custom-built W-CDMA test-bed. The data show that experimental results match well with the theoretical calibration algorithm. Also, teh study propose an efficient algorithm using the simulated annealing technique. This algorithm overcomes the requirement of initial guessing in the subspace-based approach.

Nonnegative Matrix Factorization Based Direction-of-Arrival Estimation of Multiple Sound Sources Using Dual Microphone Array (이중 마이크로폰을 이용한 비음수 행렬분해 기반 다중음원 도래각 예측)

  • Jeon, Kwang Myung;Kim, Hong Kook;Yu, Seung Woo
    • Journal of the Institute of Electronics and Information Engineers
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    • v.54 no.2
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    • pp.123-129
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    • 2017
  • This paper proposes a new nonnegative matrix factorization (NMF) based direction-of-arrival (DOA) estimation method for multiple sound sources using a dual microphone array. First of all, sound signals coming from the dual microphone array are segmented into consecutive analysis frames, and a steered-response power phase transform (SRP-PHAT) beamformer is applied to each frame so that stereo signals of each frame are represented in a time-direction domain. The time-direction outputs of SRP-PHAT are stored for a pre-defined number of frames, which is referred to as a time-direction block. Next, In order to estimate DOAs robust to noise, each time-direction block is normalized along the time by using a block subtraction technique. After that, an unsupervised NMF method is applied to the normalized time-direction block in order to cluster the directions of each sound source in a multiple sound source environments. In particular, the activation and basis matrices are used to estimate the number of sound sources and their DOAs, respectively. The DOA estimation performance of the proposed method is evaluated by measuring a mean absolute error (MAE) and the standard deviation of errors between the oracle and estimated DOAs under a three source condition, where the sources are located in [$-35{\circ}$, 5m], [$12{\circ}$, 4m], and [$38{\circ}$, 4.m] from the dual microphone array. It is shown from the experiment that the proposed method could relatively reduce MAE by 56.83%, compared to a conventional SRP-PHAT based DOA estimation method.

Fast DOA Estimation Algorithm using Pseudo Covariance Matrix (근사 공분산 행렬을 이용한 빠른 입사각 추정 알고리듬)

  • 김정태;문성훈;한동석;조명제;김정구
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.40 no.1
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    • pp.15-23
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    • 2003
  • This paper proposes a fast direction of arrival (DOA) estimation algorithm that can rapidly estimate incidence angles of incoming signals using a pseudo covariance matrix. The conventional subspace DOA estimation methods such as MUSIC (multiple signal classification) algorithms need many sample signals to acquire covariance matrix of input signals. Thus, it is difficult to estimate the DOAs of signals because they cannot perform DOA estimation during receiving sample signals. Also if the D0As of signals are changing rapidly, conventional algorithms cannot estimate incidence angles of signals exactly. The proposed algorithm obtains bearing response and directional spectrum after acquiring pseudo covariance matrix of each snapshot. The incidence angles can be exactly estimated by using the bearing response and directional spectrum. The proposed DOA estimation algorithm uses only concurrent snapshot so as to obtain covariance matrix. Compared to conventional DOA estimation methods. The proposed algorithm has an advantage that can estimate DOA of signal rapidly.