• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.1
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• pp.22-27
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• 2014

Blocking and ringing artifacts continue to be the most serious defects that occur in images and video streams compressed to low bit rates using block-based discrete cosine transform(DCT) compression standards. These artifacts contain the high frequency components near the block and the edge boundaries. Usually the lowpass filter can remove them. However, simple lowpass filter results into blur by removing important information such as edges at the same time. To overcome these problems, we propose a novel postfiltering algorithm that calculate the weight value based on the intensity similarity in the neighboring pixels and multiply this weight to the Gaussian lowpass filter coefficient. Experimental results show that the proposed technique provides satisfactory performance in both objective and subjective image quality.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.29 no.7
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• pp.500-507
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• 2018

A three-channel radio frequency (RF) monopulse receiver using a data signal with a maximum transmission rate of 274 Mbps was designed. A monopulse receiver using a broadband communication signal was designed to operate in the Ku band, and it consists of a down-conversion module and a signal-processing module. To satisfy the performance of the proposed RF monopulse receiver, a signal-processing function less than the reception sensitivity for each transmission rate according to the adaptive transmission rate is required. To minimize signal reception and mutual frequency interference of various bandwidths, two RF filters were applied. To verify the satisfaction of system requirements, an AWR Corp. simulation tool was used.

• Journal of KSNVE
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• v.5 no.4
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• pp.453-460
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• 1995

본 논문에서는 정현파 소음을 제어하기 위한 filtered-x LMS에 바탕을 둔 새로운 적응 알고리듬을 제안하였다. 이러한 알고리듬은 두개의 연속적인 계수조정 식으로, 제어기의 계수를 조정한다. 서로 독립인 각 주파수별로 처리하기 때문에 빠른 수렴을 얻을 수 있다. 두번째식은 이차경로로 인한 위상지연을 추정한다. 정현 파 신호 주파수보다 4배 이상 빠른 표본화 주파수를 선택하여 추정된 위상지연 추정 값은 $2{\pi}f_0$만큼 오차를 나타내며, 이 값은 $\pi$2보다 작다. 정현파 신호의 주파수를 알면 이러한 오차는 $2{\pi}f_0$를 더함으로써 제거할 수 있다. 이러한 방법은 위상지연이 $\pi$2보다 큰 경우 수렴속도를 증가시킨다는 사실을 실험을 통하 여 알 수 있다. 추정된 위상지연은 제어기 계수값을 조정하는데 필요한 필터링된 참조신호를 발생시키믄데 사용된다. 참조신호의 위상지연이 각 주파수 성분별로 수행 되기 때문에, 콘볼루션 연산이 생략되어 계산량을 줄일 수 있다. 또한 연속적으로 위상지연을 추정하기 때문에 시변 상황에 적용이 가능하다. 조정식의 수렴조건을 유도하였다. 제안된 알고리듬은 제어기 계수를 추정하는데 바이어스가 없으며, 위상 지연추정을 위한 수렴상수의 최대허용치는 제어기계수에 대한 수렴상수에 반비례함을 이론적으로 분석을 통해 알 수 있다. 모의실험을 통하여 제안된 알고리듬이 filtered-x LMS 알고리듬에 바탕을 둔 다른 알고리듬보다 환경변화에 우수한 성능을 보임을 알 수 있다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.630-634
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• 2005

본 논문은 신경망을 이용한 간섭 신호 제어 복합 다층 퍼셉트론에서 DS/SS 이동 통신에서 수신된 신호를 검출하는 것에 대하여 연구한다. 수신 신호가 일정한 비트율을 갖는 채널에 전송하기 위하여 신경망을 이용한 새로운 탭 가중치 갱신 제어 방법을 제안한다. 적응 횡단선 필터는 상호 심볼간 간섭을 억압하기 위해 LMS 알고리즘 사용하고, 응답과 실제 출력간의 차인 에러를 이용하여 탭 가중치 조절 메카니즘을 통해 탭 가중치를 갱신함으로서 효과적으로 간섭을 제거한다. 본 논문은 상호 심볼간 간섭을 효율적으로 억압해온 다계층 퍼셉트론 조합을 이용하여 제안된 알고리즘을 통해 탭 가중치 갱신이 보다 효율적으로 이루어질 수 있도록 한다. 시뮬레이션을 통해 평균자승 에러의 수렴 특성이 우월하다는 것을 연구한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.815-823
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• 1993

In this paper, the structure of modified multiplication-free adaptive filter(M-MADF) and convergence analysis are presented. To evaluate the performance of proposed M-MADF algorithm, fractionally spaced equalizer (FSE) is used. The input signals are quantized using DPCM and the reference signals is processed using a first-order linear prediction filter, and the outputs are processed by a conventional adaptive filter. The filter coefficients are updated using the Sign algorithm. Under the assumption that the primary and reference signals are zero mean, wide-sense stationary and Gaussian, theoretical results for the coefficient misalignment vector and its autocorrelation matrix of the filter are driven. The convergence properties of Sign. MADF and M-MADF algorithm for updating of the coefficients of a digital filter of the fractionally spaced equalizer (FSE) are investigated and compared with one another. The convergence properties are characterized by the steady state error and the convergence speed. It is shown that the convergence speed of M-MADF is almost same as Sign algorithm and is faster that MADF in the condition of same steady error. Especially it is very useful for high correlated signals.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.37-41
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• 2002

In this paper, we made shorter EDT(early decay time) of room reverberation curve using multiple-channel. The speech signal was processed inverse filtering with full-band and sub-band in the basis MINT, and then the multiple-channel adaptive filters were used LMS (Least Mean Square) and NLMS (Normalized Least Mean Square) algorithm. Experimental results, we could get 1/3 of time reduction at 20dB level in the reverberation curve using full-band NLMS when two microphones were used. Also, it is shown that the speech articulation was improved 80% from the test listeners with the speech, which was to shorten EDT by MINT in the subjective assessments using real room impulse response.

• Journal of IKEEE
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• v.13 no.2
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• pp.132-139
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• 2009

In this paper, a speech feature based double-talk detector method is proposed for an acoustic echo cancellation in hands-free communication system. The double-talk detector is an important element, since it controls the update of the adaptive filter for an acoustic echo cancellation. In previous research, the double talk detector is considered in the signal processing stage without taking the speech characteristics into account. However, in the proposed method, speech features which are used for the speech recognition is used for the discriminative features between the far-end and near-end speech. We obtained a substantial improvement over the previous double-talk detector methods using the only signal in time domain.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.278-287
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• 2003

Noise in an image degrades image quality and deteriorates coding efficiency. Recently, various edge-preserving noise filtering methods based on the nonstationary image model have been proposed to overcome this problem. In most conventional nonstationary image models, however, pixels are assumed to be uncorrelated to each other in order not to Increase the computational burden too much. As a result, some detailed information is lost in the filtered results. In this paper, we propose a computationally feasible adaptive noise smoothing algorithm which considers the nonstationary correlation characteristics of images. We assume that an image has a nonstationary mean and can be segmented into subimages which have individually different stationary correlations. Taking advantage of the special structure of the covariance matrix that results from the proposed image model, we derive a computationally efficient FFT-based adaptive linear minimum mean-square-error filter. Justification for the proposed image model is presented and effectiveness of the proposed algorithm is demonstrated experimentally.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.3
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• pp.106-115
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• 1999

본 연구에서는 충격잡음이 부가되는 비정상 신호 및 잡음 환경하에서 실시간 시지연 추정이 가능한 SLMPTDE와 ZFLMSTDE의 새로운 적응 시지연 추정 방법을 제안하였다. 본 연구에서 제안한 방법은 중간직교 척도를 바탕으로 임의의 SαS 확률과정에 강건하게 적용할 수 있도록 유도된 확률적 경사형적응 추정 알고리즘으로 구성되었으며, SαS 분포를 갖는 다양한 충격잡음을 대상으로 모의 실험하여 알고리즘의 통계적 수렴특성 및 우정 오차에 대해 분석하였으며, 기존의 LMSTDE 방법과 일정시지연의 경우와 시변시지연의 경우에 대해 실시간 시지연 추정능력을 비교, 분석하였다. 실험결과로부터, LMSTDE 방법은 α≥1.9인 가우시안 잡음에 대해서만 시지연 추정이 가능하였고 P=1로 설정한 SLMPTDE 방법은 1〈α≤2인 경우의 SαS 잡음에 대해 정확한 시지연 추정능력을 보였으며, ZFLMSTDE 방법은 0〈α≤2인 모든 경우의 잡음 환경에 대해 그 능력이 입증되었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1A
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• pp.38-47
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• 2006

In this paper, we propose a velocity-adaptive channel estimation scheme using the simple zero-forcing technique in the frequency domain. Channel estimation is performed by removing frequency components that are higher than the maximum Doppler frequency in the received signal. The proposed scheme can be extended to the combined estimation scheme for channel coefficients and mobile velocity using one FFT/IFFT module. Simulation results show that the proposed scheme outperforms conventional schemes for a wide range of mobile velocities($3{\sim}300\;Km/h$). Finally, the MSE for the proposed channel estimation scheme is analyzed.