• Journal of Communications and Networks
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• v.1 no.2
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• pp.118-133
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• 1999

In this paper we compare linear and non-linear inter-ference cancellation for systems employing code division multi-ple access (CDMA) techniques. Specifically, we examine linear and non-linear parallel interference cancellation(also called multi-stage cancellation) in relationship to other multiuser detection al-gorithms. We show the explicit relationship between parallel inter-ference cancellation and the decorrelator (or direct matrix inver-sion). This comparison gives insight into the performance of paral-lel interference cancellation (PIC) and leads to vetter approaches. We also show that non-linear PIC approaches with explicit chan-nel setimation can provide performance improvement over linear PIC, especially when using soft non-linear symbol estimates. The application of interference cancellation to non-linear modulation techniques is also presented along with a discussion on minimum mean-squared error(MMSE) symbol estimation techniques. These are shown to further improve the performance of parallel cancella-tion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.5B
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• pp.645-652
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• 2001

본 논문에서는 일반적인 코드분할 다중화 시스템을 위해 저자가 제안하였던 유사 역상관기 (Pseudo-Decorrelator)를 변형하여 WCDMA 시스템에 적합한 원근 저항성 (near-far resistant) 수신기를 설계하고, 임펄스 채널 환경에서 성능을 분석하고 있다. 설계된 수신기에서는 판단변수 (decision statistics)에 포함되어 있는 다중사용자 간섭신호를 분석하여, 송신된 신호의 각 비트에 대응되는 상호상관 행렬을 얻게 된다. 이 상호상관 행렬을 역변환 시킨 후, 기존의 수신기에서 얻어진 판단변수에 적용한다. 이렇게 해서 수신기의 성능을 향상시킬 수 있다. 제안된 수신기는 연속해서 3비트를 받은 후에 바로 검출과정 (detection process)을 수행할 수 있다는 잇점이 있다. 임펄스 채널에서의 여러 가지 시뮬레이션을 통해서 수신기의 성능이 향상됨을 확인할 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1583-1591
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• 1998

In this paper, we consider the quasi maximum likelihood(quasi-ML) detector which uses antenna arrays in asynchronous time-varing channels. It is shown that the proposed quasi-ml detector can be regarded as a beamformer followed by a decorrelator: a method based on the eigendecomposition of the correlation matrix of the inverse-filtered signal is proposed to estimate the channel vectors. We also show that the proposed algorithm estimates the channel vector within small mismatch loss in severe propagation environment through computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.340-351
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• 2012

In this paper, we proposed the stereo-10.2channel blind upmix algorithm for the enhanced 3D sound. Recently, consumers want to enjoy better sound and the use of a various of multichannel configuration has been steadily improved. Thus, upmix algorithms have been researched. However, conventional upmix algorithms have the problem that distorts the spatial information of original source. To solve this problem and enhance the spatial sound quality, we proposed front and rear channel gain adjustment and 10.2 channel upmix algorithm for each additional channel. The listening test results show that it maintains spatial information of stereo input and enhances 3D sound effects unlike other conventional upmix algorithms.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.33-42
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• 2003

In this paper a two-stage maximum-likelihood (ML) detection structure for group detection in DS/CDMA systems is presented. The first stage of the receiver is a linear filter, aimed at suppressing the effect of the unwanted (i.e., out-of-grout) users' signals, while the second stage is a non-linear block, implementing a ML detection rule on the set of desired users signals. As to the linear stage, we consider both the decorrelating and the minimum mean square error approaches. Interestingly, the proposed detection structure turns out to be a generalization of Varanasi's group detector, to which it reduces when the system is synchronous, the signatures are linerly independent and the first stage of the receiver is a decorrelator. The issue of blind adaptive receiver implementation is also considered, and implementations of the proposed receiver based on the LMS algorithm, the RLS algorithm and subspace-tracking algorithms are presented. These adaptive receivers do not rely on any knowledge on the out-of group users' signals, and are thus particularly suited for rejection of out-of-cell interference in the base station. Simulation results confirm that the proposed structure achieves very satisfactory performance in comparison with previously derived receivers, as well as that the proposed blind adaptive algorithms achieve satisfactory performance.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.5
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• pp.887-892
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• 2008

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. To analyze the convergence characteristics of the proposed algorithm, the simulations were carried out about various input signals. And we had compared the performances of convergence for this algorithm with the ones for the NLMS algorithm. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows about 5-10 dB more high SNR than the NLMS algorithm for the colored inputs.

• The Journal of the Acoustical Society of Korea
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• v.29 no.8
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• pp.509-515
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• 2010

This paper presents a new method for upmixing mono signal to stereo signal with guaranteeing high stereophonic image quality (SIQ) and large apparent source width (ASW). The proposed method consists of analysis phase and synthesis phase. In analysis phase, a mono signal is first decomposed into multiple sound sources by the use of high-rank nonnegative matrix factorization. Then the multiple sources are clustered into two groups based on tonality criterion. In synthesis phase, one group is directly fed into left and right channels while the other group is decorrelated before being fed into each channel. Subjective tests reveals that the proposed method gives listener high SIQ and large ASW with minimizing timbral distortions.