• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.4
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• pp.176-181
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• 2005

This paper presents an equal bit rate control algorithm utilizing the statistical change between the previous frame and the current frame. The previous studies on the model-based rate control have focused on the models of bit rate and distortion in types of coders, in terms of the quantization parameter. The proposed algorithm improves the typical model-based rate control by updating a model parameter instead of modeling a better model of the rate and distortion. The proposed algorithm updates this model parameter by recognizing the change in statistics between the previous frame and the current frame. We implement the proposed algorithm in MPEG-4 coders and verify its performance while comparing it to the TMN8's approach (up to 0.6dB of improvement).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.2
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• pp.11-19
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• 2003

This paper proposes an algorithm to reduce the estimation error of surface spectral-reflectance(SR) using a conventional 3-band RGB camera. In the proposed method, estimation error can be reduced by using adaptive principal components(PCs) for each color region. In order to build adaptive set of PCs, n SR populations are organized for n PC sets by using Lloyd quantizer design algorithm. Macbetch ColorCheckcer is utilized as initial representative SR values for 1485 Munsell color chips of total color population and the Munsell chips arc divided subsets and a set of corresponding adaptive PCs per each subset is organized. As a result of experiments, the proposed method showed advanced estimation performance compared to both the two 3-band PCA methods and the 5-band wiener method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4C
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• pp.505-513
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• 2004

Most video codecs including the international standards use the block-based hybrid structure for efficient compression. But for low bit-rate applications such as video transmission through wireless channels, the blocking artifacts degrade image qualify seriously. In this paper, we propose an adaptive de-blocking algorithm using characteristics of the block boundaries. Blocking artifacts contain the high frequency components near the block boundaries, therefore the lowpass filtering can remove them. However, simple lowpass filtering results into blurring by removing important information such as edges. To overcome this problem, we determine the modes depending upon the characteristics of pixels adjacent to block boundary then proper filter is applied to each area. Simulation results show that proposed method improves de-blocking performance compared to that of MPEG-4.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2
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• pp.96-100
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• 2006

Kim et al. Proposed Normalized Confidence Measure (NCM) [1-2] and it was successfully used for rejecting mis-recognized words in isolated word recognition. However their experiments were performed on the fixed word speech recognition. In this Paper we apply NCM to the domain of vocabulary independent speech recognition (VISP) and shows the rejection Performance of NCM in VISP. Specialty we Propose vector quantization (VQ) based method for overcoming the problem of unseen triphones. It is because NCM uses the statistics of triphone confidence in the case of triphone-based normalization. According to speech recognition experiments Phone-based normalization method shows better results than RLJC[3] and also triphone-based normalization approach. This results are different with those of Kim et al [1-2]. Concludingly the Phone-based normalization shows robust Performance in VISP domain.

• Journal of Broadcast Engineering
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• v.16 no.2
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• pp.216-225
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• 2011

Following the proliferation of three-dimensional video contents and displays, many terrestrial broadcasting companies have been preparing for stereoscopic 3DTV service. In terrestrial stereoscopic broadcast, it is a difficult task to code and transmit two video sequences while sustaining as high quality as 2DTV broadcast due to the limited bandwidth defined by the existing digital TV standards such as ATSC. Thus, a terrestrial 3DTV broadcasting with a heterogeneous video codec system, where the left image and right images are based on MPEG-2 and H.264/AVC, respectively, is considered in order to achieve both high quality broadcasting service and compatibility for the existing 2DTV viewers. Without significant change in the current terrestrial broadcasting systems, we propose a joint rate control scheme for stereoscopic 3DTV service based on the heterogeneous dual codec systems. The proposed joint rate control scheme applies to the MPEG-2 encoder a quadratic rate-quantization model which is adopted in the H.264/AVC. Then the controller is designed for the sum of the left and right bitstreams to meet the bandwidth requirement of broadcasting standards while the sum of image distortions is minimized by adjusting quantization parameter obtained from the proposed optimization scheme. Besides, we consider a condition on maintaining quality difference between the left and right images around a desired level in the optimization in order to mitigate negative effects on human visual system. Experimental results demonstrate that the proposed bit rate control scheme outperforms the rate control method where each video coding standard uses its own bit rate control algorithm independently in terms of the increase in PSNR by 2.02%, the decrease in the average absolute quality difference by 77.6% and the reduction in the variance of the quality difference by 74.38%.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.6
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• pp.130-136
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• 2016

HEVC (High Efficiency Video Coding) extension considers screen content as one of its main candidate sources for encoding. Among the tools already included in HEVC version 1, the technique of using transform skip mode allows transform to be skipped and to perform quantization process only. It is known to improve video coding efficiency for screen contents which are characterized to have much high frequency energy. But encoding complexity increases since its encoder should decide whether transform should be used or not in each $4{\times}4$ transform block. Based on statistical correlation between IBC (Intra block copy) and transform skip modes both of which are known effective in screen contents, this paper proposes a combined method of the fast transform skip mode decision and a modified transform skip signaling which signals transform_skip_flag at CU level as a representative transform skip signal. By simulation, the proposed method is shown to reduce encoding time of $4{\times}4$ transform blocks by about 32%.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.11
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• pp.5389-5396
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• 2012

In this paper, an optimized residual data decoder architecture is proposed to improve the performance in H.264/AVC. The proposed architecture is an integrated architecture that combined parallel inverse transform architecture and parallel inverse quantization architecture with common operation units applied new inverse quantization equations. The equations without division operation can reduce execution time and quantity of operation for inverse quantization process. The common operation unit uses multiplier and left shifter for the equations. The inverse quantization architecture with four common operation units can reduce execution cycle of inverse quantization to one cycle. The inverse transform architecture consists of eight inverse transform operation units. Therefore, the architecture can reduce the execution cycle of inverse transform to one cycle. Because inverse quantization operation and inverse transform operation are concurrency, the execution cycle of inverse transform and inverse quantization operation for one $4{\times}4$ block is one cycle. The proposed architecture is synthesized using Magnachip 0.18um CMOS technology. The gate count and the critical path delay of the architecture are 21.9k and 5.5ns, respectively. The throughput of the architecture can achieve 2.89Gpixels/sec at the maximum clock frequency of 181MHz. As the result of measuring the performance of the proposed architecture using the extracted data from JM 9.4, the execution cycle of the proposed architecture is about 88.5% less than that of the existing designs.

• Korea and Global Affairs
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• v.2 no.2
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• pp.89-130
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• 2018

This paper is to analyse the development of the U.S.-China relationship and pending issues and implications. To this end the paper is composed of 6 chapters titled instruction; the relationship between the US and China in the early and hostile confrontation period; the relationship of US-Chinese approach/normalization period and the relationship in the 1980s and 1990s; the relationship by mid-2010 since the opening of the G2 era; the US-China relations and major pending issues and implications in the era of Trump-Xi Jinping; and conclusion. The rapid growth of China over the past three decades has changed the existing US-centered international order and has triggered competition between the two countries. The United States and China have become the only countries that regularly hold strategic and economic dialogue, and the topic has also developed into a country that discusses not only bilateral relations but also global issues. The issues of US-China cooperation and conflicts encompass global issues as well as bilateral relations issues. For example, the South China Sea, the North Korean nuclear issue and the THAAD, the economic and financial order, and the Taiwan issue. It is not a matter of another country, but a problem that directly or indirectly leads to Korea's diplomacy, security and economy. In order to prevent 'Korea passing' in the US-China relationship, we need a hedging strategy that maintains and strengthens the strong ROK-US security cooperation and harmonious promotion of ROK-China economic cooperation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.