• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.28 no.12
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• pp.999-1002
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• 2017

For a target echo containing interference, it is very difficult to improve the performance of a monopulse radar with spatial domain processing, because the statistical property of interference cannot be exclusively obtained. This paper proposes a monopulse system that has a generalized sidelobe canceller(GSC) filter-based time domain processor as a preprocessor prior to conventional monopulse spatial processing. We analytically show the procedure of time-space signal processing running in the system, and assess its performance through simulation. In particular, the performance dependence on the number of taps in the main channel filter and the estimation error in Doppler frequency are assessed by comparison with those of existing systems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.05a
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• pp.742-743
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• 2010

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.30-35
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• 2006

In this paper. we Propose a frequency domain beamforming algorithm composed of inverse-filtering stages followed by a MVDR (Minimum-Variance Distortionless Response) beamformer or a GSC (Generalized Sidelobe Canceller). The proposed method is shown to require less complexity than the conventional RTF-MVDR and TF-GSC. respectively, and it is shown that the Proposed method is equivalent to the conventional RTF-MVDR and TF-GSC in optimum solution. In order to evaluate the performance of the Proposed method. speech recognition experiments are performed using the speech database recorded in a car. The Proposed method shows equal or slightly degraded Performance comparing to the conventional methods in terms of the speech recognition rate.

• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.383-391
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• 2014

In order to detect underwater acoustic signals, various SONAR array types have been developed, including towed line array SONAR system (TASS). However, the TASS suffers from performance degradation which is caused by aperture deformation during a turn, because the TASS have a long-aperture array. A parabolic array model for turning TASS have been developed to solve the degradation problem occurred during a turn. In this paper, adaptive beamforming system is developed using the parabolic TASS model to cancel interference signals. The developed beamforming system is based on generalized sidelobe canceller (GSC) structure and self-tuning adaptive algorithm.

• 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.

• 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 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.