• The Journal of the Convergence on Culture Technology
• /
• v.9 no.1
• /
• pp.471-478
• /
• 2023

Teaching and learning research is essential to prepare for the era of the 4th industrial revolution. We must strengthen the creative educational capabilities of prospective teachers by improving the curriculum and teaching-learning model of teacher training colleges. Future education should foster creative thinking based on communication and collaboration. In this study, we improved the future teaching and learning model and applied it to actual classes. In addition, we verified the effectiveness by conducting pre- and post-surveys on the capabilities of future creative teachers. The survey consists of common satisfaction questions and questions about future creative teacher capabilities. As a result of the analysis, we confirmed that the capacity of future creative teachers increased.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.23 no.6
• /
• pp.75-80
• /
• 2023

We show the design of fast invertible block transforms that can replace the DCT in future wireless and portable computing application. This is called binDCT. In binDCT, both the forward and the inverse transforms can be implemented using only binary shift and addition operation. And the binDCT inherits all desirable DCT characteristics such as high coding gain, no DC leakage, symmetric basis functions, and recursive construction. The binDCT also inherits all lifting properties such as fast implementations, invertible integer-to-integer mapping, in-place computation. Thus, this method has advantage of fast implementation for complex DCT calculations. In this paper, we present computation costs and performance analysis between DCT and binDCT using Shapiro's EZW.

• Journal of Internet Technology
• /
• v.21 no.1
• /
• pp.203-219
• /
• 2020

Compressed video sensing (CVS) is one of the 5G application of compressed sensing (CS) to video coding. Block-based residual reconstruction is used in CVS to explore temporal redundancy in videos. However, most current studies on CVS focus on random measurements without quantization, and thus they are not suitable for practical applications. In this study, an efficient ratecontrol scheme combining measurement rate and quantization for residual reconstruction in CVS is proposed. The quantization effects on CS measurements and recovery for video signals are first analyzed. Based on this, a mathematical relationship between quantitative distortion (QD), sampling rate (SR), and the quantization parameter (QP) is derived. Moreover, a novel distortion model that exhibits the relationship between QD, SR, and QP is presented, if statistical independency between the QD and the CS reconstruction distortion is assumed. Then, using this model, a rate-distortion (RD) optimized rate allocation algorithm is proposed, whereby it is possible to derive the values of SR and QP that maximize visual quality according to the available channel bandwidth.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.46 no.3
• /
• pp.68-74
• /
• 2009

In this paper, we propose an effective memory reduction algorithm to reduce the amount of reference frame buffer and memory bandwidth in video encoder and decoder. In general video codecs, decoded previous frames should be stored and referred to reduce temporal redundancy. Recently, reference frames are recompressed for memory efficiency and bandwidth reduction between a main processor and external memory. However, these algorithms could hurt coding efficiency. Several algorithms have been proposed to reduce the amount of reference memory with minimum quality degradation. They still suffer from quality degradation with fixed-bit allocation. In this paper, we propose an adaptive block-based min-max quantization that considers local characteristics of image. In the proposed algorithm, basic process unit is $8{\times}8$ for memory alignment and apply an adaptive quantization to each $4{\times}4$ block for minimizing quality degradation. We found that the proposed algorithm can obtain around 1.7% BD-bitrate gain and 0.03dB BD-PSNR gain, compared with the conventional fixed-bit min-max algorithm with 37.5% memory saving.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.51 no.12
• /
• pp.3-11
• /
• 2014

Multiple antennas can provide huge capacity gains when the transmitter knows the channel state information (CSI). Precoding is a technique that exploits CSI at the transmitter side. In this paper, an adaptive precoding scheme is proposed, called a hybrid multiple-input multiple-output (MIMO) precoding (HMP). HMP is a combination of linear and nonlinear precoding. The number of transmit antennas less than or equal to four is as same as the conventional antenna selection scheme. Therefore, the HMP scheme uses more than four transmit antennas. The good channel means that the channels must be selected to maximize the channel capacity among the given channels, and the rest channels are called bad channel. In HMP scheme, we use the nonlinear precoding in the good channels and the linear precoding in the bad channels. The well-known Tomlinson-Harashima precoding (THP) is considered as nonlinear precoding. The system throughput and MSE (minimum square error) are shown for the performance of HMP scheme compared to the conventional schemes which are BD (block diagonalization), antenna selection and THP.

• Journal of Broadcast Engineering
• /
• v.24 no.3
• /
• pp.463-471
• /
• 2019

This paper proposes a method for reducing the complexity of LIC (Local Illuminance Compensation) for bi-directional inter prediction. The LIC performs local illumination compensation using neighboring reconstruction samples of the current block and the reference block to improve the accuracy of the inter prediction. Since the weight and offset required for local illumination compensation are calculated at both sides of the encoder and decoder using the reconstructed samples, there is an advantage that the coding efficiency is improved without signaling any information. Since the weight and the offset are obtained in the encoding prediction step and the decoding step, encoder and decoder complexity are increased. This paper proposes two methods for low complexity LIC. The first method is a method of applying illumination compensation with offset only in bi-directional prediction, and the second is a method of applying LIC after weighted average step of reference block obtained by bidirectional prediction. To evaluate the performance of the proposed method, BD-rate is compared with BMS-2.0.1 using B, C, and D classes of MPEG standard experimental image under RA (Random Access) condition. Experimental results show that the proposed method reduces the average of 0.29%, 0.23%, 0.04% for Y, U, and V in terms of BD-rate performance compared to BMS-2.0.1 and encoding/decoding time is almost same. Although the BD-rate was lost, the calculation complexity of the LIC was greatly reduced as the multiplication operation was removed and the addition operation was halved in the LIC parameter derivation process.

• Journal of The Korean Association of Information Education
• /
• v.24 no.2
• /
• pp.157-165
• /
• 2020

As interest in software education has increased, discussions on teaching, learning, and evaluation method it have also been active. One of the problems of software education teaching method is that the instructor cannot grasp the content of coding in progress in the learner's computer in real time, and therefore, instructors are limited in providing feedback to learners in a timely manner. To overcome this problem, in this study, we developed a software education support system that grasps the real-time learner coding situation under block-based programming environment by applying a learning analysis technique and delivers it to the instructor, and visualizes the data collected during learning through the Hadoop system. The system includes a presentation layer to which teachers and learners access, a business layer to analyze and structure code, and a DB layer to store class information, account information, and learning information. The instructor can set the content to be learned in advance in the software education support system, and compare and analyze the learner's achievement through the computational thinking components rubric, based on the data comparing the stored code with the students' code.

• Journal of the Institute of Electronics Engineers of Korea CI
• /
• v.38 no.2
• /
• pp.33-41
• /
• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.45 no.2
• /
• pp.118-124
• /
• 2008

For forward error correction, DVB-S2, which is the digital video broadcasting forward error coding and modulation standard for satellite television, uses a system based the concatenation of BCH with LDPC inner coding. In DVB-S2 the LDPC codes are defined for 11 different code rates, which means that a DVB-S2 LDPC decoder should support multiple code rates. Seven of the 11 code rates, 3/5, 2/3, 3/4, 4/5, 5/6, 8/9, and 9/10, are regular and the rest four code rates, 1/4, 1/3, 2/5, and 1/2, are irregular. In this paper we propose a flexible decoder for the regular LDPC codes. We combined the partially parallel decoding architecture that has the advantages in the chip size, the memory efficiency, and the processing rate with Benes network to implement a DVB-S2 LDPC decoder that can support multiple code rates with a block size of 64,800 and can configure the interconnection between the variable nodes and the check nodes according to the parity-check matrix. The proposed decoder runs correctly at the frequency of 200MHz enabling 193.2Mbps decoding throughput. The area of the proposed decoder is $16.261m^2$ and the power dissipation is 198mW at a power supply voltage of 1.5V.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.39A no.4
• /
• pp.195-205
• /
• 2014

The ultra-high definition television (UHDTV) has been considered as a next generation broadcsating service. However the conventional digital terrestrial transmission system cannot afford the required transmission data rate of UHDTV, and thus adopting ultra-high order constellation, such as 4096-QAM, into the conventional DTT systems has been studied. In particular, when the ultra-high order constellation is adopted into the digital video broadcasting-2nd generation terrestrial (DVB-T2) unequal-error protection (UEP) properties of a codeword of an error correction coding and ultra-high order constellations should be properly matched by bit mapper in order to enhance the decoding performance. Because long codeword results in a heavy computational complexity to design the bit mapper, the DVB-T2 divided it into cascaded blocks, the bit interleaver and the bit-to-cell DEMUX, and there have been many researches related to each block. However, there are few published study related to design methodology of bit interleaver. In this respect, this paper proposes a design methodology of the bit interleaver and presents bit interleavers of 1024-QAM and 4096-QAM according to the proposed design algorithm. The newly designed interleavers improved the decoding performance of the error correction coding by maximally 0.6 dB SNR over both of AWGN and random fading channel.