• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.524-527
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• 2005

A society has progressed rapidly toward the highly advanced digital information age. However, noise is generated by several causes, when signal is processed. Therefore, methods for eliminating those noises have researched. There were the existing FFT(fast fourier transform) and STFT(short time fourier transform) for removing noise but it's impossible to know information about time and time-frequency localization capabilities have conflictive relationship. Therefore, for overcoming these limits, wavelet-based denoising methods that are capable of multiresolution analysis are applied to the signal processing field. However, existing threshold- and correlation-based denoising methods consider only statistical characteristics for noise, accordingly a lot of noise is acceptable as an edge and are impossible to remove AWGN and impulse noise, at the same time. Hence, in this paper we proposed wavelet-based new denoising algorithm and compared existing methods with it.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10C
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• pp.923-929
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• 2003

Turbo processing have been known as methods close to Shannon limit in the aspect of wireless multi-input multi-output (MIMO) communications similarly to wireless single antenna communication. The iterative processing can maximize the mutual effect of coding and interference cancellation, but LDPC coding has not been used for turbo processing because of the inherent decoding process delay. This paper suggests a LDPC coded MIMO system with turbo parallel space-time (Turbo-PAST) processing for high-speed wireless communications and proposes a average soft-output syndrome (ASS) check scheme at low signal to noise ratio (SNR) for the Turbo-PAST system to decide the reliability of decoded frame. Simulation results show that the suggested system outperforms conventional system and the proposed ASS scheme effectively reduces the amount of turbo processing iterations without performance degradation from the point of average number of iterations.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.3
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• pp.254-260
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• 2019

Recently, the importance of signal processing has become gradually significant, as the frequency of video media increases in various fields. However, numerous kinds of noise generated in the transmission and reception processes can possibly affect the signal information, and the noise removal is for that reason essential as a preprocessing step. In this paper, we propose an algorithm to remove the mixed noise which is composed of impulse noise and AWGN. This algorithm is used for image restoration by noise judgment for efficient noise removal in a complex noise environment, and the noise is removed by considering spatial characteristics and pixel variations. Simulation results show that unlike existing methods, the algorithm has excellent noise cancellation characteristics by minimizing both noise effects and consequently eliminating the mixed noise; for objective judgment, we compared and analyzed the data using PSNR and profile.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.3
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• pp.430-435
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• 2022

In this paper, we consider the non-orthogonal multiple access (NOMA) technique in the heterogeneous network (HetNET) consisting of a single macro base station (BS) and multiple small BSs, where the perfect successive interference cancellation is assumed for the NOMA signals. In this paper, we propose a deep learning-based user association and power allocation scheme to maximize the data rate in the NOMA-based HetNET. In particular, the proposed scheme includes the deep neural network (DNN)-based user association process for load balancing and the DNN-based power allocation process for data-rate maximization. Through the simulation assuming path loss and Rayleigh fading channels between BSs and users, the performance of the proposed scheme is evaluated, and it is compared with the conventional maximum signal-to-interference-plus-noise ratio (Max-SINR) scheme. Through the performance comparison, we show that the proposed scheme provides better sum rate performance than the conventional Max-SINR scheme.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.270-277
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• 2010

In this paper, we proposed an algorithm that improves tone quality of noisy audio signals by using ICA(Independent Component Analysis) algorithm and perceptual filters. Many algorithms have been proposed to eliminate the noise from the audio signals, such as spectral subtraction method, perceptual filter, etc. The perceptual filter uses a noise that is acquired from silent ranges in the input signal. In this case, the improvement rate of tone quality decreases if the noise energy is changed by the environmental variation in a signal frame. But the proposed method estimates a noise that is changed at each frame using ICA algorithm. The estimated noise is applied to perceptual filter. To show the performance of the proposed algorithm, several tests are performed to various input signals. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR), noise-to-mask ratio (NMR) and Degradation Category Rating (DCR) test.

• ETRI Journal
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• v.22 no.2
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• pp.11-19
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• 2000

If the right and left signals of a binaural sound recording are reproduced through loudspeakers instead of a headphone, they are inevitably mixed during their transmission to the ears of the listener. This degrades the desired realism in the sound reproduction system, which is commonly called 'cross-talk.' A 'cross-talk canceler' that filters binaural signals before they are sent to the sound sources is needed to prevent cross-talk. A cross-talk canceler equalizes the resulting sound around the listener's ears as if the original binaural signal sound is reproduced next to the ears of listener. A cross-talk canceler is also a solution to the problem-how binaural sound is distributed to more than 2 channels that drive sound sources. This paper presents an effective way of building a cross-talk canceler in which geometric information, including locations of the listener and multiple loudspeakers, is divided into angular information and distance information. The presented method makes a database in an off-line way using an adaptive filtering technique and Head Related Transfer Functions. Though the database is mainly concerned about the situation where loudspeakers are located on a standard radius from the listener, it can be used for general radius cases after a distance compensation process, which requires a small amount of computation. Issues related to inverting a system to build a cross-talk canceler are discussed and numerical results explaining the preferred configuration of a sound reproduction system for stereo loudspeakers are presented.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.1
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• pp.95-103
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• 2009

We improve the performance of WCDMA repeater by cancelling the feedback interference radio signal under the fixed point implementation. Floating-point DSP or FPGA to implement the ICS algorithm may have an disadvantage of high cost, To solve this problem, we suggest the ICS algorithm based on LMS under fixed point operation, and show the validity of our results by comparing with the floating-point results through numerical simulation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.2
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• pp.227-236
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• 2001

This paper proposes a frequency-domain partially adaptive algorithm, called a censoring algorithm, to reduce the computational complexity of the frequency domain adaptive array. The proposed censoring algorithm determines the existence of interferences in the frequency-domain at each frequency bin using a constant false alarm rate (CFAR) processor. The censoring algorithm adapts only those parts of the weights that correspond to the frequency bins expected to contain interferences. The censoring algorithm is also expanded to overcome the signal cancellation phenomenon caused by smart jammers. Accordingly, a censoring spatial smoothing, which combines the censoring algorithm with spatial smoothing, is proposed. Simulation results show that the proposed algorithms are effective in removing interferences with only part of the computational complexity of conventional algorithms yet with the same level of performance.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.6
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• pp.853-864
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• 2001

In this paper, system model is based on proposed standardization by 3GPP(Asynchronous IMT-2070 system), and we have analyzed the performance of the DS-CDMA/QPSK which cancelled self-interference, so that occurs when the received signal time is delay by multipath fading. We proposed the new scheme that designed for self-interference cancellation and the system performance is calculated in Rician fading and Rayleigh fading channel. The proposed rake receiver can be achieved a gain of about 1 [dB]∼6 [dB] more than generally rake receiver, and will be a very effective method to improve the performance of IMT-2000, and these data can be available for modem design of DS-CDMA.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.5
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• pp.2390-2395
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• 2011

For intersymbol interference (ISI) compensation from communication channels with multi-path fading and impulsive noise, a decision feedback equalizer algorithm that minimizes Euclidean distance of error probability is proposed. The Euclidean distance of error probability is defined as the quadratic distance between the probability error signal and Dirac-delta function. By minimizing the distance with respect to equalizer weight based on decision feedback structures, the proposed decision feedback algorithm has shown to have significant effect of residual ISI cancellation on severe multipath channels as well as robustness against impulsive noise.