• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.8
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• pp.1675-1685
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• 2012

DVC (Distributed Video Coding) technique plays an essential role in providing low-complexity video encoder. But, in order to achieve the better rate-distortion performances, most DVC systems need feedback channel for parity bit control. This causes the DVC-based system to have high decoding latency and becomes as one of the most critical problems to overcome for a real implementation. In order to overcome this problem and to accelerate the commercialization of the DVC applications, this paper analyzes an effect of LDPCA frame size for adaptive LDPCA frame-based parity bit request estimations. First, this paper presents the LDPCA segmentation method in pixel-domain and explains the temporal-based bit request estimation method and the spatial-based bit request estimation method using the statistical characteristics between adjacent LDPCA frames. Through computer simulations, it is shown that the better performance and fast decoding is observed specially when the LDPCA frame size is 3168 in QCIF resolution.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.5
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• pp.283-288
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• 2018

In this paper, it is required that the next generation wireless transmission system can support a large number of users without distortion of transmission signal with high data rate in various different propagation environment while using limited resources as efficiently as possible, and therefore an efficient transmission system is continuously required. Because of the large amount of data to be handled in a limited frequency band, a very complex digital modulation scheme is adopted. the linearity of the power amplifier determines the linearity of the entire communication system, and thus a linear amplifier is required. In cognitive radion systems, there is a power control issue in the relationship between primary and secondary users. This problem is solved by simulating the communication system so as to select the cognitive radio power while power control while overcoming linearity by using feed-forward PA.

• Convergence Security Journal
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• v.20 no.2
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• pp.111-121
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• 2020

Deep neural networks (DNNs) provide good performance for machine learning tasks such as image recognition and object recognition. However, DNNs are vulnerable to an adversarial example. An adversarial example is an attack sample that causes the neural network to recognize it incorrectly by adding minimal noise to the original sample. However, the disadvantage is that it takes a long time to generate such an adversarial example. Therefore, in some cases, an attack may be necessary that quickly causes the neural network to recognize it incorrectly. In this paper, we propose a fast misclassification sample that can rapidly attack neural networks. The proposed method does not consider the distortion of the original sample when adding noise. We used MNIST and CIFAR10 as experimental data and Tensorflow as a machine learning library. Experimental results show that the fast misclassification sample generated by the proposed method can be generated with 50% and 80% reduced number of iterations for MNIST and CIFAR10, respectively, compared to the conventional Carlini method, and has 100% attack rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.11A
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• pp.1146-1153
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• 2007

In this paper, we analyze the I/Q imbalances effects at the WiBro uplimk when using direct-conversion RF transceiver. If I/Q imbalance exists, the transmit signal is spread over two sbcarriers. As a result, phenomenon of performance reducing is produced. Contrary to OFDM system in which one user uses all subcarrier, symmetrical two subcarriers are assigned other users in OFDMA system. I/Q imbalances elements can't be estimated such a conventional allocation method of tiles in subchannel and compensated. In order to solve the problem, We propose a new method in order that symmetrical two subcarriers are assigned one user. If novel method is applied, we can estimate I/Q imbalances and compensate distortion received signal. As a result, we can obtain a performance similar performance when I/Q imbalances is not existed. Also, if proper detection methods are used, we get the effect of performance improvement, because of diversity gain what is happened due to combining I/Q imbalances with multi path fading channel.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.721-732
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• 2009

AT-DMB system has been developed to increase data rate up to double of conventional T-DMB in same bandwidth while maintaining backward compatibility. The AT-DMB system adopted hierarchical modulation which adds BPSK signal or QPSK signal as enhanced layer to existing DQPSK signal. The enhanced layer signal should be small enough to maintain backward compatibility and to minimize the coverage loss of existing T-DMB service area. But this causes the enhanced layer signal of AT-DMB susceptible to fading effect in transmission channel. A turbo code which has powerful error correction capability is applied to the enhanced layer signal of the AT-DMB system for compensating channel distortion. We developed the prototype AT-DMB transmitter and receiver systems for performance evaluation. LAB test for analysing the effect of constellation ratio between existing base layer signal and enhancement layer signal, was conducted and the measurement results are shown with analysis comments.

• Journal of Broadcast Engineering
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• v.20 no.6
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• pp.862-870
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• 2015

The RDO (Rate Distortion Optimization) process of HEVC results in good coding efficiency, but relatively requires much encoding time. In order to reduce the encoding time of RDO process, this paper proposes a method of fast intra prediction mode decision using DCT coefficients distributions and the existence of DCT coefficients. The proposed fast Intra coding sets the number of intra prediction mode candidates to three(3) from the RMD (Rough Mode Decision) process in HM16.0 reference SW and reduces the number of candidates one more time by investigating DCT coefficients distribution. After that, if there exists a quantized DCT block having all zero coefficient values for a specific candidate before the RDO process, the candidate is chosen without the RDO process. The proposed method reduces the encoder complexity on average 46%, while the coding efficiency is 2.1% decreased compared with the HEVC encoder.

• Journal of Broadcast Engineering
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• v.19 no.3
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• pp.355-361
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• 2014

In this paper, we propose a complexity reduction method of sample adaptive offset (SAO), which is an in-loop filter in high-efficiency video coding (HEVC). In the conventional SAO, an offset value is calculated for each coding tree block (CTB) to minimize the error between the original and reconstructed images. In order to determine the optimal offset value, all offset candidates are examined and the offset value that leads to the smallest rate-distortion cost is chosen. Thus, SAO occupies a significant amount of the computational complexity in the HEVC encoder. In the proposed method, we determine the least-used band (LUB) by considering the statistical characteristics of offset values and without processing the offset value included in the LUB. Also, in the offset value decision stage, we check only a certain number of candidates rather than all of them. Experimental results show that the proposed method reduces the encoding time by approximately 8.15% without yielding a significant loss in terms of coding efficiency.

• Journal of Broadcast Engineering
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• v.25 no.3
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• pp.305-312
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• 2020

As a demand for a new video coding standard having higher coding efficiency than the existing standards is growing, recently, MPEG and VCEG has been developing and standardizing the next-generation video coding project, named Versatile Video Coding (VVC). Many inter prediction techniques have been introduced to increase the coding efficiency, and among them, an adaptive motion vector resolution (AMVR) technique has contributed on increasing the efficiency of VVC. However, the best motion vector can only be determined by computing many rate-distortion costs, thereby increasing encoding complexity. It is necessary to reduce the complexity for real-time video broadcasting and streaming services, but it is yet an open research topic to reduce the complexity of AMVR. Therefore, in this paper, an efficient technique is proposed, which reduces the encoding complexity of AMVR. For that, the proposed method exploits a special VVC tree structure (i.e., multi-type tree structure) to accelerate the decision process of AMVR. Experiment results show that the proposed decision method reduces the encoding complexity of VVC test model by 10% with a negligible loss of coding efficiency.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.68-77
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• 2000

Line special frequency (LSF) parameters have been widely used in low bit-rate speech coding due to their efficiency for representing the short-time speech spectrum. In this paper, a new distance measure based on the masking properties of human ear is proposed for quantizing LSF parameters whereas most conventional quantization methods are based on the weighted Euclidean distance measure. The proposed method derives the perceptual distance measure from the definition of noise-to-mask ratio (NMR) which has high correspondence with the actual distortion received in the human ear and uses it for quantizing LSF parameters. In addition, we propose an adaptive bit allocation scheme, which allocates minimal bits to LSF parameters maintaining the perceptual transparency of given speech frame for reducing the average bit-rates. For the performance evaluation, we has shown the ratio of perceptually transparent frames and the corresponding average bit-rates for the conventional and proposed methods. By jointly combining the proposed distance measure and adaptive bit allocation scheme, the proposed system requires only 770 bps for obtaining 95.5% perceptually transparent frames, while the conventional systems produce 89.9% at even 1800 bps.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.15-22
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• 2012

This paper deals with the performance comparison and evaluation of blind adaptive equalizer, the PMCMA-MDD and DW-MCMA, that is used for compensation of the amplitude and phase distortion which occurs in the time dispersive channel. Basically, these algorithms are modification of MCMA cost function in order to obtain the fast convergence speed and reduced residual isi by taking the parallel and serial double structured and the combination of the concept of RCA for the updating the tap coefficient. We implements the algorithm of it and compare the recovered constellation, residual isi, MSE characteristics curve and SER in the signal to noise ratio given the time dispersive channel. As a result of simulation, the PMCMA-MDD has a good in recovered constellation than DW-MCMA. But in the SER, the DW-MCMA has a good than PMCMA-MDD.