• Title/Summary/Keyword: adaptive beamformer

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Adaptive HLMS-GSC Algorithm in Time Domain Based on Wavelets (웨이브렛에 의한 시간영역에서의 적응 HLMS-GSC 알고리듬)

  • 이정연;황석윤;홍춘표;임중수
    • Proceedings of the IEEK Conference
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    • 2002.06d
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    • pp.385-388
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    • 2002
  • This paper proposes a new GSC (Generalized Sidelobe Canceller) structure, called HLMS-GSC. Compared to Griffiths and Jim's GSC structure, the number of complex multiplication required is reduced to one half. The simulation results show that the minimum mean square errors and performance of nulling jammers by using HLMS-GSC are almost the same compared to Griffiths and Jim's GSC, although the complexity is reduced significantly. As a result, the proposed adaptive beamformer is good for real time implementation, since it has low complexity compared to previous GSC structures.

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Robust Adaptive Beamforming Using Bayesian Beam-former : A Review

  • Lee, Hyun-Seok;Yoo, Kyung-Sang;Ryu, Hee-Seob;Kwon, Oh-Kyu
    • 제어로봇시스템학회:학술대회논문집
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    • 2002.10a
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    • pp.95.6-95
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    • 2002
  • 1. Introduction 2. Basic Concepts 2.1 Signal Model 2.2. Least-Mean-Square Adaptation Algorithm 3. Minimum Mean-Square Error 4. Bayesian Beamformer References

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A Modified Robust Adaptive Beamformer for Microphone Arrays

  • Lee, Young-Ho;Choi, Su-Young;Park, Jans-Sik;Son, Kyung-Sik
    • Proceedings of the Korea Multimedia Society Conference
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    • 2003.05b
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    • pp.446-449
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    • 2003
  • The conventional GSC is inappropriate in real situation when the target signal is present. The steering vector error cancels the target signal and the target signal misadjusts the weight of the adaptive filter. To prevent the target signal cancellation, the robust GSC using the constrained adaptive filters was already proposed. However, the adaptive weight misadjustment is not settled in robust GSC. This Paper proposes a revised robust sidelobe canceller with adaptive compensator. To compensate the influence of target signal, the adaptive compensator is used in cascade. In computer simulation, we show the performance improvement by comparing the robust GSC with the proposed GSC.

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Input Signal Model Analysis for Adaptive Beamformer (적응 빔형성기의 입력신호 모델 분석)

  • Mun, Ji-Youn;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.12 no.3
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    • pp.433-438
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    • 2017
  • Containing an Angle-of-Arrival(: AOA) estimation and interference suppression techniques, an adaptive beamformer is one of core techniques for the Signal Intelligence(: SIGINT) which collect various intelligence utilizing cutting edge devices including the radar and satellite. It generates a beam with the directivity in a corresponding direction, to efficiently receive a signal from the specific direction, using antenna array. In this paper, we present the received signal model including interference signals and noise, which can be applied to an input of the signal intelligence satellite system equipped with the AOA estimation and the interference cancellation techniques, and analysis the characteristics of various signals, which can be included in the proposed received signal model. This proposed signal model can be directly applied to the performance evaluation for a variety of beamforming techniques. Also, we verify the spectrum characteristic of the presented received signal model in the frequency domain through computer simulation examples.

Speech Enhancement using Spectral Subtraction and Two Channel Beamfomer (Spectral Subtraction과 Two Channel Beamfomer를 이용한 음성 강조 기법)

  • 김학윤
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.1
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    • pp.38-44
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    • 1999
  • In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. A simulation was conducted to verify the effectiveness of the method.

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Eigenspace-Based Adaptive Array Robust to Steering Errors By Effective Interference Subspace Estimation (효과적인 간섭 부공간 추정을 통한 조향에러에 강인한 고유공간 기반 적응 어레이)

  • Choi, Yang-Ho
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37 no.4A
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    • pp.269-277
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    • 2012
  • When there are mismatches between the beamforming steering vector and the array response vector for the desired signal, the performance can be severely degraded as the adaptive array attempts to suppress the desired signal as well as interferences. In this paper, an robust method is proposed for the adaptive array in the presence of both direction errors and random errors in the steering vector. The proposed method first finds a signal-plus-interference subspace (SIS) from the correlation matrix, which in turn is exploited to extract an interference subspace based on the structure of a uniform linear array (ULA), the effect of the desired signal direction vector being reduced as much as possible. Then, the weight vector is attained to be orthogonal to the interference subspace. Simulation shows that the proposed method, in terms of signal-to-interference plus noise ratio (SINR), outperforms existing ones such as the doubly constrained robust Capon beamformer (DCRCB).

Vector Channel Simulator Design for Underwater Acoustic-based Communications

  • Kim, Duk-Yung;Kim, Yong-Deak;Lim, Yong-Kon
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.1E
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    • pp.18-24
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    • 2002
  • This paper discusses the development of an acoustic vector channel simulator for the performance analysis of an acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. The source signal (QPSK) is generated by the specified parameters. The transmitted signal generates multipath signals which have a different delay, amplitude and doppler frequency. The paper presents in details the approach to the performance analysis of an acoustic digital communication system according to the antenna structure and the various baseband signal processing techniques.

Performance Evaluation of Satellite System Based on Transmission Beamformer (송신 빔형성기 기반의 위성 시스템 구조 성능평가)

  • Mun, Ji-Youn;Hwang, Myeong-Hwan;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.13 no.4
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    • pp.713-720
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    • 2018
  • The Signal Intelligence (SIGINT) system based on Angle-of-Arrival(AOA) estimation, interference suppression, and transmission beamforming techniques is a cutting edge technology for efficiently collecting various signal information. In this paper, we present the efficient structure of a satellite system consisted of an AOA estimator, an adaptive beamformer, a signal processing and D/B unit, and a transmission beamformer, for collecting signal information. For accurately estimating AOAs of various signals, efficiently suppressing interference or jamming signals, and efficiently transmitting the collected information or data, we employ Multiple Signal Classification (MUSIC), Minimum Variance Distortionless Response (MVDR), and Minimum Mean Square Error (MMSE) algorithms, respectively. Also, we evaluate and analysis the performance of the presented satellite system through the computer simulation.

Adaptive phase monopulse for simultaneous interference cancellation in sum and difference beams (합, 차 빔 내에서 가섭 신호 동시 제거를 위한 적응 위상 모노펄스)

  • 주경환;성하종;윤대희
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.33B no.3
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    • pp.142-150
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    • 1996
  • Monopulse technique has been widely used for a bearing estimation of a desired signal. This method can estimate the bearing of the signal with only one pulse, so computational load and complexity are low and its processes are simple. But if the desired signal incidents with interferences or multiple signals exist in the beams, the method fails to estimate correct signal angle. To solve this problem, the method to use adaptive array was proposed. In conventional adaptive array methods, sum beam was formed with fixed weights and ifference beam was formed with adaptive weights determined to minimize beamformer output power with one or more linear constraints, or an algorithm to minimize each sum beam and difference beams simultaneously using adaptively formed one basic beam. Theroretical analyses and simulation results show that the performance of the proposed method is superior to that of conventional algorithms.

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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.