• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.4
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• pp.333-339
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• 2017

Transmission signal in wireless environment receives a signal in which a source signal, interference, and noise are mixed. The goal of this study is to estimate the desired signal from the received signal. In this paper, we have studied a method correctly estimating a target in spatial by modified beamformer method. The modified bemaformer uses an adaptive array antenna and perturbation matrix to obtain the optimal weight, and estimate the desired signal by radiating the beam in spatial. We estimate a desired signal of the target by improving resolution with the modified beamformer method which does not have complicated calculation amount. Through simulation, we compare and analyze the modified beamformer method and the MUSIC method with good resolution. In result of simulation, we showed that modified beamformer method has better resolution of 10degree than classical beamformer method and showed similar performance as the MUSIC method. The resolution of this paper was estimated to be about 5 degrees.

• Journal of Biomedical Engineering Research
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• v.28 no.6
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• pp.803-810
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• 2007

Although synthetic aperture focusing techniques can improve the spatial resolution of ultrasound imaging, they have not been employed in a commercial product because they require a real-time N-channel beamformer with a tremendously increased hardware complexity for simultaneous beamforming along M multiple lines. In this paper, a hardware-efficient beamformer architecture for synthetic aperture focusing is presented. In contrast to the straightforward design using NM delay calculators, the proposed method utilizes only M delay calculators by sharing the same values among the focusing delays which should be calculated at the same time between the N channels for all imaging points along the M scan lines. In general, synthetic aperture beamforming requires M 2-port memories. In the proposed beamformer, the input data for each channel is first upsampled with a 4-fold interpolator and each polyphase component of the interpolator output is stored into a 2-port memory separately, requiring 4M 2-port memories for each channel. By properly limiting the area formed with the synthetic aperture focusing, the input memory buffer can be implemented with only 4 2-port memories and one short multi-port memory.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.5 no.1
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• pp.25-36
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• 2006

This paper discusses the implementation and performance evaluation of a DSP-based digital beamformer using the Texas Instrument TMSC6711 DSP processor for smart antenna applications. Two adaptive beamforming algorithms which served as the brain for the beamformer, the Normalized Least-Mean-Square (NLMS) and the Constant Modulus Algorithms (CMA) were embedded into the processor and evaluated. Result shows that the NLMS-based digital beamformer outperforms the CMA-based digital beamformer: 1)For NLMS algorithm, the antenna steers to the direction of the desired user even at low iteration value and the suppression level towards the interferer increases as the number of iteration increase. For CMA algorithm, the beam radiation pattern slowly steers to the desired user as the number of iteration increased, but at arate slower than NLMS algorithm and the sidelobe level is shown to increases as the number of iteration increase. 2) The NLMS algorithm has faster convergence than CMA algorithm and the error convergence for CMA algorithm sometimes is subject to misadjustment.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.24 no.4
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• pp.448-458
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• 2013

In this paper, we present the performance and the experimental results of the adaptive beamformer in the radar system with the planar active array. The study of the adaptive beamformer has already been performed in several literatures, but it is difficult to find the results or examples those are implemented in the actual radar system. Here we employ the adaptive beamformer to the practical radar system with subarray architecture. The performance of beamformer will be demonstrated by modeling and simulation and finally the far-field experimental results.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.520-525
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• 2009

The conventional studies about an adaptive beamformer assumed that the interference signals are stationary, so they used time-average of signals or Least Mean Squares. However, these methods showed low performance of canceling the non-stationary interferences. In this paper, the MAFF-RLS algorithm is developed in order to cancel non-stationary interferences, and the GSC structure using this algorithm is proposed. Furthermore, the performance of the MAFF-RLS beamformer is verified by simulation using MATLAB. This simulation results show the performance of the proposed beamformer is better than that of the SMI and the conventional RLS beamformer.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.40 no.4
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• pp.342-350
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• 1991

This paper describes an adaptive beamformer which improves the frequency characteristics of the AMNOR system. The proposed beamformer uses a bank of filters to calculate the frequency degradation in each subband. In the process ofdesigning the beamformer, the filter coefficients are updated to minimize the power of the difference between a fictitious desired signal and the output under the permissible degradation value in each subband. Simulation results demonstrating the improved performances of the proposed method are presented.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.199-202
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• 2002

This paper introduces the near field shading beamformer using widely known Chebyshev and Hanning window in the field of digital signal processing. The proposed shading beamformer improves the estimation of range as well as azimuth angle of targe residing in near field. A series of sensor weighting values are calculated from the FFT operation of given shading functions in time domain. This paper verifies the performance of the focused beamformer having the proposed shading sensor weights which are used to detect the range of target. Throughout computer simulations this paper exploits the performance improvement of the proposed shading beamformer as varying the frequency band of the received radiated signal along the non-uniform array.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.2 s.332
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• pp.49-59
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• 2005

An adaptive hybrid beamformer is proposed to improve the reception performance of the advanced television system committee (ATSC) digital television (DTV) in a mobile environment. Dynamic multipaths and Doppler shifts severely degrade the reception performance of the ATSC DTV receiver. Accordingly, a hybrid beamformer, called a Capon and least mean square (CLMS) beamformer, is presented that uses direction of arrival (DOA) information and the least mean square (LMS) beamforming algorithm. The proposed CLMS algerian efficiently removes dynamic multipaths and compensates for the phase distortion caused by Doppler shifts in mobile receivers. After the CLMS beamformer is operated, the subsequent use of an equalizer removes any residual multipath effects, thereby significantly improving the performance of DTV receivers. The performances of the proposed CLMS, Capon, and LMS beamformersare compared based on computer simulations. In addition, the overall performance of the CLMS beamformer followed by an equalizer is also considered.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.543-543
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• 2010

The objective of this research is estimating the location of interested sound source by using the similarity between a beamformer output in time domain and the candidate signal. The waveform of beamformer output at the location of sound source is similar with the waveform emitted by that source. To estimate the location of sound source by using this feature, we define quantified similarity between candidate signal and beamformer output. The candidate signal describes the signal which is generated by interested source. In this paper, similarity is defined by four methods. The two methods use time vector comparison, and the other two methods use time-frequency map or linear prediction coefficients. To figure out the results and performance of localization by using similarities, we demonstrate two conditions. The one is when two pure tone sources exist and the other condition is when several bird sounds exist. As a consequence, inner product with two time-vectors and structural similarity with spectrograms can estimate the locations of interest sound source.

• The Journal of the Acoustical Society of Korea
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• v.36 no.2
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• pp.100-107
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• 2017

The focused MVDR (Minimum Variance Distortionless Response) can be applied for source localization in near field. However, if the number of sensors are increased, it requires a large amount of calculation to obtain the inverse of the covariance matrix. In this paper we propose a focused MVDR method using that beam space is formed from output of far field beamformer at the subarray. The performances of the proposed method was evaluated by simulation. As a result of simulation, the proposed method has the higher spatial resolution performance then the conventional delay-and-sum beamformer.