• The Journal of the Acoustical Society of Korea
• /
• v.19 no.8
• /
• pp.54-62
• /
• 2000

Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.17 no.2
• /
• pp.121-131
• /
• 1992

In this paper, three methods are studied to enhance the performance of SBC ( Sub-Band Coding )schemes for voice signal at the medium bit rate between 12 kbps and If; kbps, and adaptive postfilteritng using human auditory characteristics Is (Bone at the decoder out put. First, GQMF(Generalized Quadrature Mirror Filter ) Is used instead of QME'((Quadrature MirrorFiltcr ) to have better performance. Second, by adaptive bit allocation to each sub-band, speech quality is enhanced and valuable rate ceding If possible. Third, corriparlson study oS thr: coder performance using APCM(Adaptive Pulse Code ModulatioTi) and ADPCM( Adaptive Differentiai Pulse Code Modulatiori) , Indicates that SB AfCM performance better than the other. Adaptive postfiltering at the decoder output enhances the quality of the coded speech. The two proposed postfiltering methods decrease the noise sufficiently at the expense of the low computational load.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.38 no.4
• /
• pp.442-452
• /
• 2001

In this paper, we proposed a postprocessing algorithm for quantization effects reduction in block coded images using the block classification and adaptive filtering. The proposed method consists of classification, adaptive inter-block filtering, and intra-block filtering. First, each block is classified into one of seven classes based on the characteristics of 8$\times$8 DCT coefficients. Then each block boundary is filtered by adaptive inter-block fitters according to the block classification. finally for blocks which are classified into edge block, intra-block filtering is performed. Experimental results show that the proposed method gives better results than the conventional methods from both a subjective and an objective viewpoint.

• The Journal of the Acoustical Society of Korea
• /
• v.6 no.2
• /
• pp.15-18
• /
• 1987

Multiplication free adaptive digital filtering algorithms are discussed. The proposed. The proposed algorithm uses delta modulation digital filter and the relevant filter weights are updated using the SIGN algorithms to realize an adaptive digital filter without multiplication operations. It is shown that the resulting algorithm can be implemented using simple up/down counting operations. The convergence characteristics of the proposed adaptive digital filtering algorithm and .others are investigated for a system identification problem.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2010.07a
• /
• pp.294-296
• /
• 2010

일반적으로, RGB 영상의 높은 주파수 영역은 잡음으로 인해 색평면 간 서로 낮은 상관도를 가지고 있기 때문에 이러한 고주파수 성분은 색평면 간 예측의 효율을 저하시키는 원인이 된다. 본 논문에서는 RGB 비디오 코딩에서 색평면 간 예측의 효율을 높이기 위해 기저 색평면을 적응적으로 필터링 하는 방법을 제안한다. 색평면 간 상관도에 따라 적응적으로 기저 색평면을 필터링함으로써 색평면 간 예측 성능을 높일 수 있었다. 본 논문에서 제안하는 알고리즘을 통해 우리는 H.264/AVC High 4:4:4 Intra Profile에 비해 평균 14.71%의 비트율 감소와 0.93dB의 PSNR 향상 결과를 얻을 수 있었다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2009.11a
• /
• pp.53-56
• /
• 2009

최근 HD 방송의 실현과 함께 HDTV가 빠르게 상용화되면서 고화질 비디오를 더 효율적으로 압축하기 위한 기술 개발이 활발하게 진행되고 있다. 최신 표준 비디오 압축 방식인 H.264/AVC에서는 고정 6탭 필터를 사용하여 참조 영상을 보간하고 움직임 예측을 수행하는데, 이의 압축 효율을 향상시키기 위하여 고정 필터를 개선한 비분리 적응형 보간 필터가 개발되었다. 하지만 비분리 적응형 보간 필터는 압축률뿐만 아니라 계산량도 증가하는 단점이 있기 때문에 2차원 필터를 1차원 필터의 연결로 모델링 한 분리 적응형 보간 필터가 개발되었고, 그 결과 압축률은 비슷하게 유지하면서 계산량을 훨씬 줄일 수 있게 되었다. 분리 적응형 보간 필터는 1차원 필터 모델링을 할 때, 수평방향의 필터링 수행 후 수직방향 필터링을 하기 때문에 이를 통해 만들어진 보간 영상은 원 영상의 수평방향에 대한 특성을 더 많이 반영하게 된다. 따라서 수직방향으로 더 높은 주파수 특성을 갖는 영상의 경우에는 효율이 떨어지게 된다. 이를 고려하여 본 논문에서는 영상의 수직방향 주파수 특성을 더 많이 반영할 수 있는 보간 필터를 추가함으로써 영상의 주파수 특성에 따라 보간 필터를 적응적으로 선택하는 향상된 분리 적응형 보간 필터를 제안한다. 제안한 알고리즘을 이용할 경우 기존의 분리 적응형 보간 필터에 비해 움직임 예측 및 보상이 더 정확하게 이뤄질 수 있으며, 부호화 효율이 향상됨을 확인할 수 있다.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.7
• /
• pp.45-55
• /
• 1999

In this paper, we used the WLMS(Wavelet domain Least Mean Square) adaptive filter based on the wavelet transform to cancel grain noise. Usually, grain noise occurs in changes of the crystalline structure of metals in high temperature environment. It makes the detection of flaw difficult. The WLMS adaptive filtering algorithm establishes the faster convergence rate by orthogonalizaing the input vector of adaptive filter as compared with that of LMS adaptive filtering algorithm in time domain. We implemented the WLMS adaptive filter by using the delayed version of the primary input vector as the reference input vector and then implemented the CA-CFAR(Cell Averaging- Constant False Alarm Rate) threshold estimator. CA-CFAR threshold estimator enables to detect the flaw and back echo signals automatically. Here, we used the output signals of adaptive filter as its input signal. To Cow the statistical characteristic of ultrasonic signals corrupted by grain noise, we performed run test. The results showed that ultrasonic signals are nonstationary signal, that is, signals whose statistical properties vary with time. The performance of each filter is appreciated by the signal-to-noise ratio. After LMS adaptive filtering in time domain, SNR improves to about 2-3㏈ but after WLMS adaptive filtering in wavelet domain, SNR improves to about 4-6㏈.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.4 no.2
• /
• pp.407-416
• /
• 2000

Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.47 no.5
• /
• pp.127-135
• /
• 2010

In motion adaptive temporal noise reduction filtering used for reducing video noises, the strength of motion adaptive temporal filtering should be carefully controlled according to temporal movement. This paper presents a motion adaptive temporal filtering scheme based on least-square training. Each pixel is classified to a specific class code according to temporal movement, and then, an iterative least-square training method is applied for each class code to find optimal filtering coefficients. The iterative least-square training is an off-line procedure, and the trained filter coefficients are stored in a lookup table (LUT). In actual noise reduction filtering operation, after each pixel is classified by temporal movement, simple filtering operation is applied with the filter coefficients stored in the LUT according to the class code. Experiment results show that the proposed method efficiently reduces video noises without introducing blurring.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.26 no.3
• /
• pp.443-448
• /
• 2022

This paper presents a subband affine projection algorithm with variable step size to improve convergence performance in adaptive filtering applications with long adaptive filters and highly correlated input signals. The proposed algorithm can obtain fast convergence speed and small steady-state error by using different step sizes for each adaptive sub-filter in the subband structure to which polyphase decomposition and noble identity are applied. The step size derived to minimize the mean square error of the adaptive filter at each update time shows better convergence performance than the existing algorithm using a variable step size. In order to confirm the convergence performance of the proposed algorithm, which is superior to the existing algorithm, computer simulations are performed for mean square deviation(MSD) for AR(1) and AR(2) colored input signals considering the system identification model.