• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.945-951
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• 2007

Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.10
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• pp.1386-1391
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• 2018

The radar in the moving platform is interfered by the mainlobe clutter as well as the altitude clutter that is received from sidelobe. The altitude clutter is relatively short range compared to mainlobe clutter and therefore enters the radar with a strong signal. As these clutters are major reason making the probability of false alarm high, it is required to suppress both altitude clutter and mainlobe clutter. In this paper, It is proposed the clutter suppression method consisted of two pulse canceller to suppress the clutters being two frequency area in moving platform. It is analyzed the correlation of output signals according to the use of pulse canceller and provided the structure of staggered pulse canceller considered the correlation. Finally, it shows that altitude clutter and mainlobe clutter are suppressed by proposed staggered pulse canceller using the simulation.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1187-1190
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• 1998

In this paper we propose a new efficient generalized sidelobe canceller(GSC) algorithm, using the efficient frequency domain LMS algorithm, which has much less weights to update. We only update part of all the weights according to magnitude of each frequency bin. So, the new proposed GSC algorithm, which is called censored GSC(C-GSC), can greatly reduce the computational complexity.

• 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.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.389-392
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• 2002

This paper propose a new GSC (Generalized Sidelobe Canceller) structure, called HFLMS-GSC. The number of complex multiplication required is reduced to one half compared to FLMS-GSC. The simulation results show that mean square error converging and jamming signal removing characteristics are almost the same compared to FLMS-GSC, although the complexity is reduced significantly. As a result, the proposed structure is good for real time implementation, since it has low complexity compared to previous GSC structures.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.144-155
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• 1998

The array weights that will maximize the SNR for any type of noise environment are determined by the function of the antenna design configuration and the directions of receiving target and interference signals. The conventional SLCs(sidelobe cancellers) using the SNR maximization perform worst from the saturation of the receiving system of main channel when the main antenna has pattern with high gain at the arrival angle of strong interference. In this paper, the new SLC is accomplished by using two independent antenna architecture. Main antenna is implemented with adaptive nulling, which is used for rejecting high-power interference primarily. Auxiliary antenna is realized with adaptive array for receiving interference signal to be suppressed completely, which has a characteristics of sufficient gain for every direction. The new SLC is implemented with above both antennas. We show that the new SLC, which consists of the adaptive nulling main antenna and the adaptive array auxiliary antenna, is useful in reducing the effect of strong interference like jammer, because the adaptive nulling at main antenna prevents its receiver and signal processor for saturation by strong interference. The proposed SLC has improved SNR over the conventional SLCs. The improved SNR at sidelobe region is typically more than 7 dB for a given test signal. Moreover, it improves the SNR of about 20 dB under strong interference at mainlobe.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.1
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• pp.61-68
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• 2015

Noise reduction is necessary to compensate for the degradation of recognition performance by various types of noises. Among many noise reduction techniques using microphone array, generalized sidelobe canceller (GSC) has been widely applied to reduce nonstationary noises. The performance of GSC is directly affected by its adaptation mode controller (AMC). That is, accurate target speech detection is essential to guarantee the sufficient noise reduction in pure noise intervals and the less distortion in target speech intervals. Thus, this paper proposes an improved AMC design technique in which the power ratio of the output of fixed beamforming to that of blocking matrix is calculated frequency bandwise and probabilistically modeled by mixture Gaussians for each class. Experimental results show that the proposed algorithm outperforms conventional AMCs in receiver operating curves (ROC) and output SNRs.

• Phonetics and Speech Sciences
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• v.1 no.3
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• pp.95-102
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• 2009

In this paper, an efficient target speech detection algorithm is proposed for the performance improvement of multi-input speech enhancement. Using the normalized cross correlation value between two selected channels, the proposed algorithm estimates the probabilistic distribution function of the value from the pure noise interval. Then, log-likelihoods are calculated with the function and the normalized cross correlation value to detect the target speech interval precisely. The detection results are applied to the generalized sidelobe canceller-based algorithm. Experimental results show that the proposed algorithm significantly improves the speech recognition performance and the signal-to-noise ratios.

• 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.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.