• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.1
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• pp.21-40
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• 2018

In this paper, we propose the distributed robust beamforming design scheme in cognitive two-way amplify-and-forward (AF) relay networks with imperfect channel state information (CSI). Assuming the CSI errors follow a complex Gaussian distribution, the objective of this paper is to design the robust beamformer which minimizes the total transmit power of the collaborative relays. This design will guarantee the outage probability of signal-to-interference-plus-noise ratio (SINR) beyond a target level at each secondary user (SU), and satisfies the outage probability of interference generated on the primary user (PU) above the predetermined maximum tolerable interference power. Due to the multiple CSI uncertainties in the two-way transmission, the probabilistic constrained optimization problem is intractable and difficult to obtain a closed-form solution. To deal with this, we reformulate the problem to the standard form through a series of matrix transformations. We then accomplish the problem by using the probabilistic approach based on two sorts of Bernstein-type inequalities and the worst-case approach based on S-Procedure. The simulation results indicate that the robust beamforming designs based on the probabilistic method and the worst-case method are both robust to the CSI errors. Meanwhile, the probabilistic method can provide higher feasibility rate and consumes less power.

• The Journal of the Acoustical Society of Korea
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• v.11 no.3
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• pp.53-60
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• 1992

Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1471-1479
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• 1999

This paper presents an adaptive beamforming algorithm for an MC-CDMA system with an adaptive array antenna. By employing an antenna array at the receiver of an MC-CDMA system, the performance of an MC-CDMA system, which is known to be effective for high data rate transmission due to its robustness to multipath fading and its simplicity for using a simple one-tap equalizer, is shown to be significantly improved. The proposed algorithm for adaptive beanforming in an MC-CDMA system is derived by (1) calculating the error signals between the pilot symbols of desired user and the received pilot signals in frequency domain, (2) transforming the frequency-domain error signals into time-domain error signals, (3) updating the filter coefficients of the adaptive beamformer in the direction of minimizing the MSE. Convergence behavior and performance improvement of the proposed approach are demonstrated through computer simulation by applying it to the conventional MC-CDMA system.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.5
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• pp.786-795
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• 1999

To detect the precise of arrival of target signal in real ocean environments, Inverse beamformnig(IBF) solutions to the Inverse beamforming integral equation are surveyed theoretically and the performance properties of the IBF are analyzed with simulations. IBF-Cardioid beamforming algorithm is proposed for port/starboard discrimination and the performance gains are studied with simulations. It is shown that IBF has a 3 dB array noise gain advantage over CBF under ideal conditions. This 3 dB array noise gain advantage is proven by theocratical studies and simulations. This array noise gain advantage leads to a minimum detectable level advantage for IBF output compared with CBF output. The fact that the IBF beamwidth is narrower than the CBF beamwidth by a factor of 0.68 proves the performance of detection and spatial resolution improvement. Comparing the simulation results of IBF-Cardioid beamforming and Conventional Cardioid beamforming, it is shown that IBF-Cardioid beamformer have performance enhancement in minimum detection level, detection accuracy and resolution.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.2
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• pp.451-456
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• 2016

Because GPS signals are weak, system jamming is a real and present danger. This can happen when the receiver is subjected to intentional or unintentional interference by a transmitter. If the jamming signal is strong enough, the receiver can be operated to take corrective action automatically. Current methods to protect GPS receiver from jamming condition are based on spatial filtering. In this paper, the beamforming as referred to in signal processing technique used in arrays for directional signal reception was suggested and analyzed for anti-jamming. In order to change the directionality of the array when receiving a jamming signal, a beamformer can control the signal at each sensor. Therefore, cutoff angle ${\theta}$ was measured in the opposite direction of the jammer. GPS signals are only processed when the antenna element is within inside the cutoff angle. As a result, GPS positioning can be used in condition under cutoff angle $30^{\circ}$.

• Journal of Communications and Networks
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• v.15 no.4
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• pp.383-397
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• 2013

In this paper, the sensitivity of massive antenna arrays and digital beamforming to radio frequency (RF) chain in-phase quadrature-phase (I/Q) imbalance is studied and analyzed. The analysis shows that massive antenna arrays are increasingly sensitive to such RF chain imperfections, corrupting heavily the radiation pattern and beamforming capabilities. Motivated by this, novel RF-aware digital beamforming methods are then developed for automatically suppressing the unwanted effects of the RF I/Q imbalance without separate calibration loops in all individual receiver branches. More specifically, the paper covers closed-form analysis for signal processing properties as well as the associated radiation and beamforming properties of massive antenna arrays under both systematic and random RF I/Q imbalances. All analysis and derivations in this paper assume ideal signals to be circular. The well-known minimum variance distortionless response (MVDR) beamformer and a widely-linear (WL) extension of it, called WL-MVDR, are analyzed in detail from the RF imperfection perspective, in terms of interference attenuation and beamsteering. The optimum RF-aware WL-MVDR beamforming solution is formulated and shown to efficiently suppress the RF imperfections. Based on the obtained results, the developed solutions and in particular the RF-aware WL-MVDR method can provide efficient beamsteering and interference suppressing characteristics, despite of the imperfections in the RF circuits. This is seen critical especially in the massive antenna array context where the cost-efficiency of individual RF chains is emphasized.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.49-56
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• 2017

This paper proposes a method to improve the left right discrimination performance by eliminating the imaginary target based on the frequency features of the beam pattern for bow array. The beamwidth of the imaginary target is wider than that of the real target. If an azimuth axis is considered as a time axis, the real and the imaginary targets can be assumed as high and low frequencies, respectively. To eliminate the imaginary target which has a low frequency component, we design a cut-off frequency of the High Pass Filter (HPF) using the back-lobe imaginary beamwidth. The real target is estimated by eliminating the imaginary target by applying HPF to the entire power of the beamformer output. Computer simulations show that the proposed method can increase the left right discrimination performance above 8 dB on average.

• The Journal of the Acoustical Society of Korea
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• v.39 no.4
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• pp.255-262
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• 2020

In modern passive sonar systems, a towed array sensor is used to minimize the effects of own ship noise and to get a higher SNR. The thin and long towed array sensor can be guided in a non-linear form according to the maneuvering of tow-ship. If this change of the array shape is not considered, the performance of beamformer may deteriorate. In order to properly beamform the elements in the array, an accurate estimate of the array shape is required. Various techniques exist for estimating the shape of the linear array. In the case of a method using a heading sensor, the estimation performance may be degraded due to the effect of heading sensor noise. As means of removing this potential error, weighted polynomial fitting technique for estimating array shape is developed here. In order to evaluate the performance of proposed method, we conducted computer simulation. From the experiments, it was confirmed that the proposed method is more robust to noise than the conventional method.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.439-451
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• 2021

We propose a deep learning-based beamformer combined with spectral subtraction for continuous speech recognition operating in noisy environments. Conventional beamforming systems were mostly evaluated by using pre-segmented audio signals which were typically generated by mixing speech and noise continuously on a computer. However, since speech utterances are sparsely uttered along the time axis in real environments, conventional beamforming systems degrade in case when noise-only signals without speech are input. To alleviate this drawback, we combine online beamforming algorithm and spectral subtraction. We construct a Continuous Speech Enhancement (CSE) evaluation set to evaluate the online beamforming algorithm in noisy environments. The evaluation set is built by mixing sparsely-occurring speech utterances of the CHiME3 evaluation set and continuously-played CHiME3 background noise and background music of MUSDB. Using a Kaldi-based toolkit and Google web speech recognizer as a speech recognition back-end, we confirm that the proposed online beamforming algorithm with spectral subtraction shows better performance than the baseline online algorithm.