• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.433-438
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• 1999

This paper describes Active Noise Cancellation/Control(ANC) method that removes the information of the unnecessary noise and doesn't remove the informations of the necessary noise(warning sound, operating sound etc.) for the induced noise of the mechanical system. In this paper, the noise source Is axial fan, and the Feedback Active Noise control method that can effectively control BPF generated from the axial fan is used, and the Filtered-X LMS algorithm for adaptive algorithms is used. The experiments are executed for two case(propagating noise in the duct, emission noise for exterior free field). The part to be removed is BPF noise, and the band-pass filter not to effect to the other frequencies is used. Also, to investigate the effect of the noise reduction for human, we are compared with the results that are controlled for using Loudness before and after. As a results, we are certified that the BPF is decreased only and frequencies outside of BPF are not affected, and we acquire the reduction effects of 6.7 dB Loudness Level, then the frequency to be removed is controlled. Therefore, we can be certified that sound pressure as well as loudness can be effectively decreased for human sound quality

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.18 no.2 s.117
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• pp.177-182
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• 2007

This paper presents the InGaP/GaAs HBT differential LC VCO with low phase noise performance for adaptive feedback interference cancellation system(AF-lCS). The VCO is verified with enhanced tank structure including filtering technique. The output tuning range for proposed VCO using asymmetric inductor and symmetric capacitors withlow pass filtering technique is 207 MHz. The output powers are -6.68 including balun and cable loss. The phase noise of this VCO at 10 kHz, 100 kHz and 1 MHz are -102.02 dBc/Hz, -112.04 dBc/Hz and -130.40 dBc/Hz. The VCO is designed within total size of $0.9{\times}0.9mm^2$.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.4
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• pp.3-8
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• 2015

In this paper, we propose a blind equalization algorithm based on the error signal linearly combined a constellation-matched error (CME) and a non-constellation-matched error (non-CME). The new error signal was designed to include the non-CME term for reaching initial convergence and the CME term for improving intersymbol interference (ISI) performance of output signals, and it controls the error terms through a combining factor. By controlling the error terms, it generates an appropriate error signal for equalization process and improves convergence speed and ISI cancellation performance compared to those of conventional algorithms. In the simulation for 64-QAM and 256-QAM signals under the multipath channel and additive noise conditions, the proposed method was superior to CMA and CMA+DD concurrent equalization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10C
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• pp.967-983
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• 2003

In wireless spread-spectrum communication systems utilizing PN (pseudo noise) sequences, a variety of noise sources from the channel affect the data reception performance. Among them, in this paper we are concerned with the narrowband interference that may arise from the use of the spectral bands overlapped by the existing narrowband users or the intentional jammers as in military communication. The effect of this interference can be reduced to some extent at the receiver with the PN demodulation by processing gain. It is known, however, that when the interferers are strong, the reduction cannot be sufficient and thereby requiring the extra use of narrowband interference cancellers (NIC's) at the receivers. A class of adaptive NIC's are studied here based on different two cost functions. One is the chip mean-squared error (MSE) computed prior to the PN demodulation and used in the conventional cancellers. Since thses conventional cancellers should be operated at the chip rate, the computational requirements are enormous. The other is the symbol MSE computed after the PN demodulation in which case the weights of the NIC's can be updated at a lot lower symbol rate. To compare the performance of these NIC's, we derive a common measure of performance, i.e., the symbol MSE after the PN demodulation. The analytical results are verified by computer simulation. As a result, it is shown that the cancellation capability of the symbol-rate NIC's are similar or better than the conventional one while the computational complexity can be reduced a lot.

• The Journal of the Korea Contents Association
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• v.6 no.12
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• pp.89-95
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• 2006

The performance of blind source separation(BSS) using independent component analysis (ICA) declines significantly in a reverberant environment. A post-processing method proposed in this paper was designed to remove the residual component precisely. The proposed method used modified NLMS(normalized least mean square) filter in frequency domain, to estimate cross-talk path that causes residual cross-talk components. Residual cross-talk components in one channel is correspond to direct components in another channel. Therefore, we can estimate cross-talk path using another channel input signals from adaptive filter. Step size is normalized by input signal power in conventional NLMS filter, but it is normalized by sum of input signal power and error signal power in modified NLMS filter. By using this method, we can prevent misadjustment of filter weights. The estimated residual cross-talk components are subtracted by non-stationary spectral subtraction. The computer simulation results using speech signals show that the proposed method improves the noise reduction ratio(NRR) by approximately 3dB on conventional FDICA.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.4
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• pp.274-280
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• 2009

In order to estimate the accurate location of a user, Global Positioning system (GPS) requires at least four satellites. Since a conventional despreader operate for an GPS signal of interest, we need multiple despreaders for detecting multiple GPS signals. In this paper, we introduce the extension of the recently proposed system consisting of a null despreader, a conventional despreader, multi-stage CM (constant modulus) array, for the multiple GPS signals, and present the mathematical expression of the signal-to-interference-and-noise ratio (SINR). The extended system does not require the exact information of the direction of arrival (DOA) to suppress the directional interferences. We present the computer simulation to demonstrate the interference suppression performance of the proposed system for multiple GPS signals.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.7
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• pp.589-602
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• 2009

The phenomenon degrading reception performance of DVB broadcasting signal is caused by collocation of plural communication modules in the GSM-900 cellular phone with a built-in DVB-H receiver. This paper offers how to improve performance of the DVB-H receiver by applying digital filter that is using DSP techniques to cancel GSM signal interfering to the DVB-H receiver which is collocated in a single user's GSM terminal. In the demodulation part of the DVB-H receiver, the DVB reception performance can be greatly improved by removing interference components of GSM signal, which is flowed with DVB signal, to the noise cancellation filtering algorithm using difference of characteristics between GSM modulated signal and OFDM signal.