• Proceedings of the IEEK Conference
• /
• 2002.07a
• /
• pp.188-191
• /
• 2002

To reduce the bit-rate of video sequences by removing temporal redundancy, motion estimation techniques have been developed. However, the high computational complexity of the problem makes such techniques very difficult to be applied to high-resolution applications in a real time environment. For this reason, low computational complexity motion estimation algorithms are viable solutions. If a priori knowledge about the motion of the current block is available before the motion estimation, a better starting point for the search of n optimal motion vector on be selected and also the computational complexity will be reduced. In this paper, we present an adaptive block matching algorithm based on temporal correlations of consecutive image frames that defines the search pattern and the location of initial starting point adaptively to reduce computational complexity. Experiments show that, comparing with DS(Diamond Search) algorithm, the proposed algorithm is about 0.1∼0.5(㏈) better than DS in terms of PSNR and improves as much as 50% in terms of the average number of search points per motion estimation.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.53 no.12
• /
• pp.36-41
• /
• 2016

LDPC code has beend applied in recent communication standards, such as Wi-Fi, WiGig, 10GBased-T Ethernet as a forward error correction code. However, LDPC code is required a large amount of computational complexity due to large iterations and block lengths for high performances. To solve this problem, various research has been continously performed for reducing computational complexity. In this paper, we propose AFC algorithm to deactive the variable and check node for reduce the computational complexity.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.16 no.4
• /
• pp.1330-1350
• /
• 2022

As a preprocessing operation of transmitter antennas, the hybrid precoding is restricted by the limited computing resources of the transmitter. Therefore, this paper proposes a novel hybrid precoding that guarantees the communication efficiency with low complexity and a fast computational speed. First, the analog and digital precoding matrix is derived from the maximum eigenvectors of the channel matrix in the sub-connected architecture to maximize the communication rate. Second, the extended power iteration (EPI) is utilized to obtain the maximum eigenvalues and their eigenvectors of the channel matrix, which reduces the computational complexity caused by the singular value decomposition (SVD). Third, the Aitken acceleration method is utilized to further improve the convergence rate of the EPI algorithm. Finally, the hybrid precoding based on the EPI method and the Aitken acceleration algorithm is evaluated in millimeter-wave (mmWave) massive multiple-input and multiple-output (MIMO) systems. The experimental results show that the proposed method can reduce the computational complexity with the high performance in mmWave massive MIMO systems. The method has the wide application prospect in future wireless communication systems.

• Journal of the Institute of Convergence Signal Processing
• /
• v.19 no.2
• /
• pp.42-46
• /
• 2018

Adaptive signal processing is quite important in various signal and communication environments. In adaptive signal processing methods since the least mean square(LMS) algorithm is simple and robust, it is used everywhere. As the step is varied in the variable step(VS) LMS algorithm, the fast convergence speed and the small excess mean square error can be obtained. Various variable step LMS algorithms are researched for better performances. But in some of variable step LMS algorithms the computational complexity is quite large for better performances. The fixed step LMS algorithm with a low computational complexity merit and the variable step LMS algorithm with a fast convergence merit are combined in the proposed sporadic step algorithm. As the step is sporadically updated, the performances of the variable step LMS algorithm can be maintained in the low update rate using Fibonacci sequence. The performances of the proposed variable step LMS algorithm are proved in the adaptive equalizer.

• Proceedings of the Korea Contents Association Conference
• /
• 2006.11a
• /
• pp.499-502
• /
• 2006

H.264/MPEG4 Advanced Video coding joint standard needs deblocking filter of decoder. We propose a better deblocking algorithm ensuring picture quality even if it is low bit-rate and bandwidth in MPEG-4 video. The complexity diminishes in proposed deblocking algorithm because it uses only simple shift, addition and comparison. We handle dividing into complexity area, medium area and simple area after counting boundary intensity of mask block to identify presence of block effects. As a result of experiment, we make certain of that block effects reduces in proposed deblocking algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.11 no.4
• /
• pp.1987-2001
• /
• 2017

Low-density parity-check (LDPC) codes have attracted a great attention because of their excellent error correction capability with reasonably low decoding complexity. Among decoding algorithms for LDPC codes, the min-sum (MS) algorithm and its modified versions have been widely adopted due to their high efficiency in hardware implementation. In this paper, a self-adaptive MS algorithm using the difference of the first two minima is proposed for faster decoding speed and lower power consumption. Finding the first two minima is an important operation when MS-based LDPC decoders are implemented in hardware, and the found minima are often compressed using the difference of the two values to reduce interconnection complexity and memory usage. It is found that, when these difference values are bounded, decoding is not successfully terminated. Thus, the proposed method dynamically decides whether the termination-checking step will be carried out based on the difference in the two found minima. The simulation results show that the decoding speed is improved by 7%, and the power consumption is reduced by 16.34% by skipping unnecessary steps in the unsuccessful iteration without any loss in error correction performance. In addition, the synthesis results show that the hardware overhead for the proposed method is negligible.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2007.10a
• /
• pp.332-335
• /
• 2007

A real-time frame-layer rate control algorithm with a transmission buffer is proposed for minimizing video quality variation. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize variation in distortion between frames. In order to reflect the buffer status, we use well-known PID control method. Computational complexity of PID control is very low, so the proposed algorithm is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better PSNR performance than the existing rate control method.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.11 no.11
• /
• pp.2009-2014
• /
• 2007

A novel method of minimizing video quality variation is proposed for a real-time frame-layer rate control algorithm with a transmission buffer. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize variation in distortion between frames. In order to reflect the buffer status, we use well-known PID control method. Computational complexity of PID control is very low, so the proposed algorithm is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better PSNR performance than the existing rate control method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.3C
• /
• pp.322-328
• /
• 2009

This paper proposes new simplified sum-product (SSP) decoding algorithm to improve BER performance for low-density parity-check codes. The proposed SSP algorithm can replace multiplications and divisions with additions and subtractions without extra computations. In addition, the proposed SSP algorithm can simplify both the In[tanh(x)] and tanh-1 [exp(x)] by using two quantization tables which can reduce tremendous computational complexity. Moreover, the simulation results show that the proposed SSP algorithm can improve about $0.3\;{\sim}\;0.8\;dB$ of BER performance compared with the existing modified sum-product algorithms.

• The Transactions of the Korean Institute of Electrical Engineers D
• /
• v.51 no.2
• /
• pp.86-88
• /
• 2002

An estimation method for the characteristic parameters in the low frequency signal is proposed in this paper. A low frequency signal is assumed to be modulated or distorted by high frequency terms. The algorithm proposed in this paper is designed to select set of local maximums in a successive manner, hence it is denoted as the iterative peak picking(IPP) algorithm. The IPP algorithm is operating in the time domain and is using only the comparison operation between two neighboring samples. Therefore, its computational complexity is very low and the delay caused by the computation is negligible, which make the real-time operation possible with economic hardware. The proposed algorithm is verified on the pitch estimation of speech signal and blood pulse estimation.