• Proceedings of the KIEE Conference
• /
• 2004.07d
• /
• pp.2652-2654
• /
• 2004

홈 네트워크나 가로등 제어에 가장 적합한 솔루션 중 하나가 전력선 통신기술이다. 전력선 통신의 사용 주파수 대역은 $50kHz{\sim}450kHz$으로, 이 주파수 대역은 잡음이 매우 많이 존재한다. 또한 시공간에 따른 변화의 요인이 많이 존재하기 때문에 채널 모델링이 매우 어려운 실정이며, 이를 극복할 수 있는 방안으로 영교차점을 이용한 전력선 모뎀을 설계하고자 한다. 변조 방식은 Chirp 심볼을 사용하고, 주파수 대역은 $100kHz{\sim}400kHz$를 사용하였다. 전력선 채널로부터 정보를 획득하기 위해서 효율적인 수신기 구조를 채택하였다. 즉 채널 상황에 적응이 가능하고, 상관에 필요한 필터의 계수를 들어오는 심볼로부터 구하는 방법으로, 다음에 들어오는 심볼과의 상관값을 구하여 정보 신호를 판별하는 형태로 구성하였다. 영교차점을 기준으로 신호를 송수신하는 형태를 취했으며, 영교차점 기준에 들어오는 잡음은 chirp의 대역 확산 특성에 의해서 억제될 수 있다. 이런 기술에 의해서 견실한 전력선 통신 모뎀구현이 가능하였다.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.18 no.4
• /
• pp.790-796
• /
• 2014

Identification of communication channels is a problem of important current theoretical and practical concerns. Recently proposed solutions for this problem exploit the diversity induced by antenna array or time oversampling. The method resorts to an adaptive filter with a linear constraint. In this paper, an approach is proposed that is based on decomposition. Indeed, the eigenvector corresponding to the minimum eigenvalue of the covariance matrix of the received signals contains the channel impulse response. And we present an adaptive algorithm to solve this problem. Proposed technique shows the better performance than one of existing algorithms.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.3
• /
• pp.245-250
• /
• 2009

In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.10
• /
• pp.2303-2309
• /
• 1997

This paper presents a convergence analysis of the stocastic gradient adaptive algorithm based on the least mean absolute third (LMAT) error criteriohn. Under the assumption that the signals involved are zero-mean, wide-sense sateionaryand gaussian, a set of nonlinear difference equations that characterizes the mean and mean-squared behavior of the algorithm is derived. Computer simulation resutls show fairly good agreements between the theoetical and empirical behaviors of the algorithm.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2004.11a
• /
• pp.197-200
• /
• 2004

디지털 TV 전송 방식중의 하나인 ATSC 8-VSB 시스템의 등화기는 훈련신호가 존재하는 구간에서 LMS 알고리즘을 사용하는 DFE 적옹 등화기가 사용된다. 그러나 LMS 알고리즘은 그 수렴속도가 느리고 수렴 후 오차 수준이 다른 적응 알고리즘에 비해 높다는 단점이 있다. 본 논문에서는 LMS 알고리즘을 사용하는 DFE의 오차 수준을 낮추기 위한 선형 등화기 구조의 전 처리부(pre-processor)를 사용하여 필터 수렴 후의 DFE의 오차수준을 기존의 DFE보다 낮추었으며 제안된, DFE 구조의 성능을 컴퓨터 모의 실험을 통해 분석하였다.

• Journal of the Korean Institute of Telematics and Electronics T
• /
• v.35T no.2
• /
• pp.48-55
• /
• 1998

The main drawback of filtered-X LMS(FXLMS) algorithm for the ANC of broadband noises is its low convergence speed when the filtered reference signals are strongly correlated, producing a large eigenvalue ratio in correlation matrix. This correlation can be caused either by autocorrelation of the signals of the reference sensors, or by coupling between the error path which introduces intercorrelation in the filtered reference signals. In this paper, we introduce a transform domain FXLMS(TD-FXLMS) algorithm that has a high convergence speed by orthogonal transform's decorrelation properties.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.1
• /
• pp.27-31
• /
• 2012

In this paper, we constructed an array of speaker system with directivity by using FXLMS(Filtered-X LMS) algorithm and confirmed its directivity. The front $0^{\circ}$ characteristics of the controlled speaker was suppressed by interfering it with the control signal produced with filter coefficients optimized with respect to the $180^{\circ}$ characteristics of the rear speakers. The directivity of the array of rear speakers was measured and the damping effect of the signal from the front speaker array was confirmed. The frequency characteristics and directivity was investigated by using the adaptive filter coefficients on damping, the damping on the control point was verified in all the frequency range. In 100Hz, 200Hz, 1000Hz regime, the damping effect was observed in the range of front $60^{\circ}{\sim}100^{\circ}$.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.15 no.4
• /
• pp.360-367
• /
• 2004

In this paper, we have implemented the digital signal processing system for intruder detection using speckle pattern variation in multi-me optical fiber with hypersensitive and high fidelity. The performance of the implemented system was evaluated by experiments. In order to improve the system performances we applied the adaptive digital filter. In experimental results we could see 96 % intruder detection and 90 % man/car discrimination probability.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.9C
• /
• pp.1337-1344
• /
• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.8
• /
• pp.1774-1781
• /
• 1997

In this paper, we propose a full digital frequency and phase locked loop for CATV and HDTV receivers adopting VSB modulation. The CATV and HDTV receivers proposed by the Grand-Alliance in USA are ultilizing analog signal processing technology for carrier recovery. By the way, it is not a good architecture for the development of single chip ASIC operating in digital domain. To solve this problem while improving the performance, we first down convert the received r.f. signal to a near baseband signal for a low-rate AD converter and then we use digital signal processing techniques. The proposed system has the frequency pull-in range of -200 KHz +2.50 KHz. Moreover, it has the ability of adaptive loop bandwidth control according to the amount of frequency offset to improve the acquisition time while reducing the phase noise.