• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.10 no.4
• /
• pp.333-339
• /
• 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 the Korean Institute of Telematics and Electronics B
• /
• v.30B no.8
• /
• pp.16-22
• /
• 1993

As a method of separating the signal arriving from a desired direction in the presence of noise and interfering signals, a linearly constrained adaptive beamformer based on the escalator (Gram-Schmidt) structure is presented. The weights are chosen to minimize local output power subject to the response constraint. The performances of the proposed escalator adaptive beamformer are compared with those of cascade adaptive beamformer via computer simulations.

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

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

• Proceedings of the IEEK Conference
• /
• 2002.06d
• /
• pp.199-202
• /
• 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 Biomedical Engineering Research
• /
• v.28 no.6
• /
• pp.803-810
• /
• 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.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.4A
• /
• pp.399-406
• /
• 2011

Adaptive beamformers, which utilize a priori information on the arrival angle of the desired signal. suppress interferences while maximizing their gains in the desired signal direction. However, if there exist errors in the direction information, they can suffer from severe performance degradation since the desired signal is treated as an interference. A robust adaptive beamforming method is presented which exploits the signal-blocking structure of the Duvall beamformer. The proposed method finds an interference signal space directly from correlations of received signals and then obtains a weight vector such that it is orthogonal to the space. Applying the weight vector to two sub arrays which consist of one less sensors than the original uniform lineal array (ULA), the beamformer efficiently estimates the arrival angle of the desired signal. Its computational complexity is lower than existing methods, which require matrix inversion or eigendecomposition.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.2
• /
• pp.52-57
• /
• 2001

The passive ranging estimation techniques using a focused beamformer have been studied under the water. It is well known that the passive ranging estimation method using a focused beamformer is excellently evaluated. Among these, the passive ranging sonar is known to have a good performance under low signal-to-noise. ratio. However, its performance is degraded in multi-source environments. In this paper, we proposed the technique using dual focused beamformers to estimate the range. And when the sampling frequency is low, it is very difficult to steer the focused beam to the desired direction, as a result of this, the low performance occurs because of a distorted beam pattern. In this paper, we study the effect of sampling rate on passive ranging by using focused beamformer. And we verified the performance of the proposed method via computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.48 no.4
• /
• pp.160-168
• /
• 2011

A Frequency domain beamforming technique is widely used in sonar systems with a large number of beams and sensors. In the battlefield environment requiring real-time signal processing, it is needed to optimize the computational complexity of the spectrum computation to implement an efficient and fast frequency domain beamformer. So, in this paper, we proposed the pruned-GSFFT (pruned generalized sliding fast Fourier transform) as a new spectrum computation method. The proposed method help to reduce the computational complexity of the real-time partial spectrum computation by eliminating the redundancy between consecutive input samples and skipping the regardless frequency bands. Also the characteristics of the proposed pruned-GSFFT method and its computational complexity are compared to those of previous FFT algorithms.

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