• Title/Summary/Keyword: Broadband Beamforming

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A Broadband Microstrip Array Antenna for 3G Smart Antenna System Testbed

  • Rashid, Zainol Abidin Abdul;Islam, Mohammad Tariqul;Jiunn, Ng Kok
    • Journal of The Institute of Information and Telecommunication Facilities Engineering
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    • v.7 no.1
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    • pp.41-58
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    • 2007
  • A compact and broadband $4{\times}1$ array antenna was developed for 3G smart antenna system testbed. The $4{\times}1$ uniform linear array antenna was designed to operate at 1.885 to 2.2GHz with a total bandwidth of 315MHz. The array elements were based on the novel broadband L-probe fed inverted hybrid E-H (LIEH) shaped microstrip patch, which offers 22% size reduction to the conventional rectangular microstrip patch antenna. For steering the antenna beam, a commercial variable attenuator (KAT1D04SA002), a variable phase shifter (KPH350SC00) with four units each, and the corporate 4-ways Wilkinson power divider which was fabricated in-house were integrated to form the beamforming feed network. The developed antenna has an impedance bandwidth of 17.32% ($VSWR{\leq}1.5$), 21.78% ($VSWR{\leq}2$) with respect to center frequency 2.02GHz and with an achievable gain of 11.9dBi. The design antenna offer a broadband, compact and mobile solution for a 3G smart antenna testbed to fully characterized the IMT-2000 radio specifications and system performances.

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A Relay Selection and Power Allocation Scheme for Cooperative Wireless Sensor Networks

  • Qian, Mujun;Liu, Chen;Fu, Youhua;Zhu, Weiping
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.8 no.4
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    • pp.1390-1405
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    • 2014
  • This paper investigates optimal relay selection and power allocation under an aggregate power constraint for cooperative wireless sensor networks assisted by amplify-and-forward relay nodes. By considering both transmission power and circuit power consumptions, the received signal-to-noise ratio (SNR) at the destination node is calculated, based on which, a relay selection and power allocation scheme is developed. The core idea is to adaptively adjust the selected relays and their transmission power to maximize the received SNR according to the channel state information. The proposed scheme is derived by recasting the optimization problem into a three-layered problem-determining the number of relays to be activated, selecting the active relays, and performing power allocation among the selected relays. Monte Carlo simulation results demonstrate that the proposed scheme provides a higher received SNR and a lower bit error rate as compared to the average power allocation scheme.

Wideband DOA Estimation with FDFIB Network (FDFIB Network를 이용한 광대역 DOA 추정)

  • Zhou, Weiwei;Wang, Yisu;Jang, Woo-Jin;Koh, Jin-Hwan
    • Proceedings of the IEEK Conference
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    • 2006.06a
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    • pp.251-252
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    • 2006
  • In this paper, we presented an extension of the broadband DOA estimation method using frequency-domain frequency-invariant beamforming (FDFIB). The technique uses FDFIB instead of conventional frequency invariant beamforming (FIB) methods. And different narrowband DOA estimation methods, MUSIC, ESPRIT, and MPM, are used respectively. A comparison is made to demonstrate that the FDFIB-MPM not only offers a better resolution than the FDFIB-MUSIC, FDFIB-ESPRIT, but also it is computationally very efficient.

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Quantitative Evaluation of the Performance of Monaural FDSI Beamforming Algorithm using a KEMAR Mannequin (KEMAR 마네킹을 이용한 단이 보청기용 FDSI 빔포밍 알고리즘의 정량적 평가)

  • Cho, Kyeongwon;Nam, Kyoung Won;Han, Jonghee;Lee, Sangmin;Kim, Dongwook;Hong, Sung Hwa;Jang, Dong Pyo;Kim, In Young
    • Journal of Biomedical Engineering Research
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    • v.34 no.1
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    • pp.24-33
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    • 2013
  • To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

Direction finding based on Radon transform in frequency-wavenumber domain with a sparse array (주파수-파수 스펙트럼과 라돈변환을 이용한 희소 배열 기반 방위추정 기법 연구)

  • Choi, Yong Hwa;Kim, Dong Hyeon;Kim, J.S.
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.2
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    • pp.168-176
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    • 2019
  • When an array receives a signal with a frequency higher than the design frequency, there is an ambiguity in beamforming due to spatial aliasing. In order to overcome this problem, Abadi proposed frequency-difference beamforming. However, there is a constraint that the minimum frequency bandwidth is required according to the value of the difference frequency. In this paper, we propose a method to find the direction of the target signal with spatial aliasing based on the frequency-wavenumber spectrum combined with Radon transform. The proposed method can estimate the direction of the target without ambiguities when the signal has nonnegligible bandwidth. We tested the algorithm by simulating a broadband signal and verified the results with the frequency-difference beamforming method using SAVEX15 (Shallow Water Acoustic Variability EXperiment 2015)'s shrimp noise data.

A Broadband FIR Beamformer for Underwater Acoustic Communications (수중음향통신을 위한 광대역 FIR 빔형성기)

  • Choi, Young-Chol;Lim, Yong-Kon
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.10 no.12
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    • pp.2151-2156
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    • 2006
  • Beamforming for underwater acoustic communication (UAC) is affected by the broadband feature of UAC signal, which has relatively low currier frequency as compared to the signal bandwidth. The narrow-band assumption does not hold good in UAC. In this paper, we discuss a broadband FIR beamformer for UAC using the baseband equivalent way signal model. We consider the broadband FIR beamformer for QPSK UAC with carrier frequency 25kHz and symbol rate 5kHz. Array geometry is a uniform linear way with 8 omni-directional elements and sensor spacing is the half of the carrier wavelength. The simulation results show that the broadband n beamformer achieves nearly optimum signal to interference and noise ratio (SINR) and outperforms the conventional narrowband beamformer by SINR 0.5dB when two-tap FIR filter is employed at each sensor and the inter-tap delay is a quarter of the symbol interval. The broadband FIR beamformer performance is more degraded as the FIR filter length is increased above a certain value. If the inter-tap delay is not greater than half of the symbol period, SINR performance does not depend on the inter-tap delay. More training period is required when the inter-tap delay is same as the symbol period.

Adaptive GSC using Subband Filter Structure in Broadband Beamforming (서브밴드 필터구조를 이용한 광대역 적응 GSC)

  • Lee, Seung-Youl;Lee, Young-Jin;Yoo, Kyung-Yul
    • Proceedings of the KIEE Conference
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    • 2002.07d
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    • pp.2592-2594
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    • 2002
  • 본 논문에서는 GSC(Generalized Sidelobe Canceller)를 기초로 새로운 부밴드 광대역 적응 빔포밍 구조를 제안하였다. 일반적으로 여러개의 필터계수를 갖는 광대역 빔포밍에서는 그 필터길이가 커짐에 따라 많은 계산량을 필요로 하고 그 성능이 감소한다는 단점이 있었다. 이러한 단점을 보완하기 위해 부밴드 필터구조를 이용함으로써 전밴드 필터구조에서보다 더 낮은 계산량과 그 pre-whitening 효과로 그 성능이 향상되었다. 부밴드 필터뱅크 구조에서 광대역 적응 빔포밍이 수행될 때 NLMS(Normalized Least Mean Squares) 적응 알고리즘을 이용하여 GSC의 수렴성능을 검증하였고, 각각의 부밴드 적응필터에서 MSE를 독립적으로 최소화시키는 적응 메카니즘을 사용하여 추정하였다. 모의실험을 통하여 제안한 부밴드 필터구조가 전밴드 구조에서보다 수렴성능이 더 우수함을 검증하였다.

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Implementation of Dual-Mode Channel Card for SDR-based Smart Antenna System (SDR기반 스마트 안테나 시스템을 위한 듀얼 모드 채널 카드 구현)

  • Kim, Jong-Eun;Choi, Seung-Won
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.12A
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    • pp.1172-1176
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    • 2008
  • In this paper, we describe the implementation and performance of a dual-mode Software Define Radio (SDR) smart antenna base station system. SDR technology enables a communication system to be reconfigured through software downloads to the flexible hardware platform that is implemented using programmable devices such as Digital Signal Processors (DSPs), Field Programmable Gate Arrays (FPGAs), and microprocessors. The presented base station channel card comprises the physical layer (pHY) including the baseband modem as well as the beamforming module. This channel card is designed to support TDD High-Speed Downlink Packet Access (HSDPA) as well as Wireless Broadband Portable Internet (WiBro) utilizing the SDR technology. We first describe the operations and functions required in WiBro and TDD HSDPA. Then, we explain the channel card design procedure and hardware implementation. Finally, we evaluate WiBro and TDD HSDPA performance by simulation and actual channel-card-based processing. Our smart antenna base-station dual-mode channel card shows flexibility and tremendous performance gains in terms of communication capacity and cell coverage.

Sound Source Detection Technique Considering the Effects of Source Bandwidth and Measurement Noise Correlation (소음원 대역폭과 측정잡음의 상관관계를 고려한 소음원 탐지기법)

  • 윤종락
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.2
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    • pp.86-92
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    • 2001
  • Various array processing techniques to identify the noise source position or bearing have been developed. Typical array processing techniques which are based on time delay between received signals at two sensors, are classified as conventional beamforming, correlation function and NAH (Near-Field Acoustic Holography) techniques which have their own characteristics with respect to application field and signal processing method. In this study, correlation function technique which could be applied for broadband noise source detection, is adopted and the effective detection technique is proposed considering the effects of source bandwidth and measurement noise correlation of noise sources. The validity of the Proposed technique is evaluated using the 3-dimensional nonlinear any which does not give 3-dimensional Position or bearing ambiguity

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