• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.10 no.6
• /
• pp.308-316
• /
• 1985

A new implementation of Block Adaptive filter in frequency domain is presented in this paper. Block digital filtering involves the calculation of a block or finite set of filter out put from a block of input values. A fast convergence algorithm of block adaptive filter is developed using Gordar theory and compared with the performance results of Satio algorithm and BLMS algorithm. Form the result we can be shown that the convergence state of given algorithm is not only faster than BLMS algorithm but also the resulting convergence error is less than the convergence error of Satio algorithm.

• Proceedings of the IEEK Conference
• /
• 2001.06d
• /
• pp.123-126
• /
• 2001

Block filter implementation technique for uniform filter banks is uniform in this paper. By applying block filter into decimation and interpolation filters, it is shown that down and up samplers are cancelled out in respective liters. Furthermore by applying block filters into uniform filter banks, significant reduction for computational complexity is achieved since prototype filter can be shared in each channel implementation. Also, it is shown that proposed implementation is a reconfigurable structure in terms of order variation.

• Journal of Broadcast Engineering
• /
• v.17 no.6
• /
• pp.1048-1060
• /
• 2012

In this paper, an adaptive temporal and spatial filter (ATSF) based on mean square error (MSE) estimation is proposed. ATSF is a block based de-noising algorithm. Each noisy block is selectively filtered by a temporal filter or a spatial filter. Multi-hypothesis motion compensated filter (MHMCF) and bilateral filter are chosen as the temporal filter and the spatial filter, respectively. Although there is no original video, we mathematically derivate a formular to estimate the real MSE between a block de-noised by MHMCF and its original block and a linear model is proposed to estimate the real MSE between a block de-noised by bilateral filter and its original block. Finally, each noisy block is processed by the filter with a smaller estimated MSE. Simulation results show that our proposed algorithm achieves substantial improvements in terms of both visual quality and PSNR as compared with the conventional de-noising algorithms.

• Proceedings of the IEEK Conference
• /
• 2002.07a
• /
• pp.377-380
• /
• 2002

When it calculates the optimum price of filter coefficient, the many operation quantity is necessary. Is like that the real-time control is difficult and the hardware embodiment expense is big. The case which does not know advance information of input signal or the case where the statistical nature changes with change of surroundings environment is necessary the adaptive filter. Every hour to change a coefficient automatically and system in order to reach to the condition of optimum oneself, the fact that is the adaptive filter. When it does not the quality of input signal or it does not know the environment of surroundings every hour changing, it does not emit not to be, in order to collect, the fact that is the adaptive filter. The case of the Acoustic Echo Canceler does thousands filter coefficients in necessity. It reduces a many calculation quantity to respect, it uses the IIR filter from hour territory. Also it uses the block adaptive filter which has a block input signal and a block output signal. The former there is a weak point where the stability discrimination is always demanded. Consequently, The block adaptive filter is researched plentifully. This dissertation planned the block MADF adaptive filter used to MADf algorithm.

• 제어로봇시스템학회:학술대회논문집
• /
• 2005.06a
• /
• pp.1435-1438
• /
• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.17 no.10
• /
• pp.1109-1116
• /
• 1992

Mathematical morphology, theoretical foundation for morphological filter, is very efficient for the analysis of the geometrical characteristics of signals and systems and is used as a predominant tool for smoothing the data with noise. In this study, H/W design of morphological filter is implemented to process the gray scale dilation and the erosion in the same circuit and to choose the maximum and minimum value by a selector, resulting in their education of the complexity of the circuit and an architecture for parallel processing. The structure of morphological filter consists of the structuring-element block, the image data block, the control block, the ADD block, the MIN/MAX block, etc, and is designed on an one-chip for real time operation.

• ETRI Journal
• /
• v.4 no.3
• /
• pp.18-29
• /
• 1982

In a block adaptive filtering procedure, the filter coefficients are adjusted once per each output block while maintaining performance comparable to that of widely used LMS adaptive filtering in which the filter coefficients are adjusted once per each output data sample. An efficient implementation of block adaptive filter is possible by means of discrete transform technique which has cyclic convolution property and fast algorithms. In this paper, the block adaptive filtering using Fermat Number Transform (FNT) is investigated to exploit the computational efficiency and less quantization effect on the performance compared with finite precision FFT realization. And this has been verified by computer simulation for several applications including adaptive channel equalizer and system identification.

• ETRI Journal
• /
• v.32 no.3
• /
• pp.380-389
• /
• 2010

This paper presents a new edge-protection algorithm and its very large scale integration (VLSI) architecture for block artifact reduction. Unlike previous approaches using block classification, our algorithm utilizes pixel classification to categorize each pixel into one of two classes, namely smooth region and edge region, which are described by the edge-protection maps. Based on these maps, a two-step adaptive filter which includes offset filtering and edge-preserving filtering is used to remove block artifacts. A pipelined VLSI architecture of the proposed deblocking algorithm for HD video processing is also presented in this paper. A memory-reduced architecture for a block buffer is used to optimize memory usage. The architecture of the proposed deblocking filter is verified on FPGA Cyclone II and implemented using the ANAM 0.25 ${\mu}m$ CMOS cell library. Our experimental results show that our proposed algorithm effectively reduces block artifacts while preserving the details. The PSNR performance of our algorithm using pixel classification is better than that of previous algorithms using block classification.

• Proceedings of the IEEK Conference
• /
• 2003.07b
• /
• pp.827-830
• /
• 2003

This paper proposes an area-efficient FIR filter architecture for sampling rate conversion of hi-fi audio data. Sampling rate conversion(SRC) block converts audio data sampled at 96KHz down to 48KHz sampled data and vice versa. 63-tap FIR filter coefficients have been synthesized that gives 100dB stop band attenuation and 5.2KHz transition bandwidth. Time-shared filter architecture requires only one multiplier and accumulator for 63-tap filter operation. This results in huge hardware saving of up to 10~19 times smaller compared with traditional FIR structure.

• 제어로봇시스템학회:학술대회논문집
• /
• 2000.10a
• /
• pp.240-240
• /
• 2000

This paper presents two algorithms based on the concept of the frequency domain adaptive filter(FDAF). First the frequency domain recursive least squares(FRLS) algorithm with the overlap-save filtering technique is introduced. This minimizes the sum of exponentially weighted square errors in the frequency domain. To eliminate discrepancies between the linear convolution and the circular convolution, the overlap-save method is utilized. Second, the sliding method of data blocks is studied Co overcome processing delays and complexity roads of the FRLS algorithm. The size of the extended data block is twice as long as the filter tap length. It is possible to slide the data block variously by the adjustable hopping index. By selecting the hopping index appropriately, we can take a trade-off between the convergence rate and the computational complexity. When the input signal is highly correlated and the length of the target FIR filter is huge, the FRLS algorithm based on the block processing technique has good performances in the convergence rate and the computational complexity.