• ETRI Journal
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• v.42 no.2
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• pp.258-271
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• 2020

A correlation exists between luminance samples and chrominance samples of a color image. It is beneficial to exploit such interchannel redundancy for color image compression. We propose an algorithm that predicts chrominance components Cb and Cr from the luminance component Y. The prediction model is trained by supervised learning with Laplacian-regularized least squares to minimize the total prediction error. Kernel principal component analysis mapping, which reduces computational complexity, is implemented on the same point set at both the encoder and decoder to ensure that predictions are identical at both the ends without signaling extra location information. In addition, chrominance subsampling and entropy coding for model parameters are adopted to further reduce the bit rate. Finally, luminance information and model parameters are stored for image reconstruction. Experimental results show the performance superiority of the proposed algorithm over its predecessor and JPEG, and even over JPEG-XR. The compensation version with the chrominance difference of the proposed algorithm performs close to and even better than JPEG2000 in some cases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.10
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• pp.850-857
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• 2013

Turbo equalizer system is a method which can improve performance through a combination of the equalizer and decoder. The turbo equalizer has been mainly used a MAP equalizer. However, this turbo equalizer has a disadvantage that has a high computational complexity. To overcome the disadvantage and to improve efficiency of bandwidth, blind turbo equalization system is proposed. blind turbo equalization system has low equalization performance than conventional turbo equalization system. To circumvent this problem, we adapt the beamforming method based on the MUSIC algorithm. we confirmed that the proposed method improves the equalization performance.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.348-356
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• 2005

In this paper, we propose a fast bitrate reduction transcoding method to convert a bitstream coded by H.264 into a lower bitrate H.264 bitstream. Block mode informations and motion vectors generated by H.264 decoder are used for probability-based block mode determination in the proposed transcoding method. And the motion vector reuse and motion vector refinement process are applied in the proposed transcoding. In the experiment results, the proposed methods achieves approximately 40 times improvement in computation complexity compared with the cascaded pixel domain transcoding, while the PSNR(Peak Signal to Noise Ratio) is degraded with only $0.1\~0.3$ dB.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.1
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• pp.426-445
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• 2012

Conventional rate control (RC) schemes for H.264 video coding usually regulate output bit rate to match channel bandwidth by adjusting quantization parameter (QP) at fixed full frame rate, and the passive frame skipping to avoid buffer overflow usually occurs when scene changes or high motions exist in video sequences especially at low bit rate, which degrades spatial-temporal quality and causes jerky effect. In this paper, an active content adaptive frame skipping scheme is proposed instead of passive methods, which skips subjectively trivial frames by structural similarity (SSIM) measurement between the original frame and the interpolated frame via motion vector (MV) copy scheme. The saved bits from skipped frames are allocated to coded key ones to enhance their spatial quality, and the skipped frames are well recovered based on MV copy scheme from adjacent key ones at the decoder side to maintain constant frame rate. Experimental results show that the proposed active SSIM-based frameskip scheme acquires better and more consistent spatial-temporal quality both in objective (PSNR) and subjective (SSIM) sense with low complexity compared to classic fixed frame rate control method JVT-G012 and prior objective metric based frameskip method.

• Journal of Korea Multimedia Society
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• v.20 no.5
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• pp.731-739
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• 2017

Distributed compressed video sensing (DCVS) is a framework that integrates both compressed sensing and distributed video coding characteristics to achieve a low complexity video sampling. In DCVS schemes, motion estimation & motion compensation is employed at the decoder side, similarly to distributed video coding (DVC), for a low-complex encoder. However, since a simple BCS-SPL algorithm is applied to a residual arising from motion estimation and compensation in conventional MC-BCS-SPL (motion compensated block compressed sensing with smoothed projected Landweber) scheme, the reconstructed visual qualities are severly degraded in Wyner-Ziv (WZ) frames. Furthermore, the scheme takes lots of iteration to reconstruct WZ frames. In this paper, the conventional MC-BCS-SPL algorithm is improved to be operated in more effective way in WZ frames. That is, first, the proposed algorithm calculates a correlation coefficient between two reference key frames and, then, by selecting adaptively the reference frame, the residual reconstruction in pixel domain is performed to the conventional BCS-SPL scheme. Experimental results show that the proposed algorithm achieves significantly better visual qualities than conventional MC-BCS-SPL algorithm, while resulting in the significant reduction of the decoding time.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.1
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• pp.37-43
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• 2023

Compressed still image or video bitstreams require protection from channel errors in a wireless channel. Embedded Zerotree Coding(EZW), SPIHT could have provided unprecedented high performance in image compression with low complexity. If bit error is generated by dint of wireless channel transmission problem, the loss of synchronization on between encoder and decoder causes serious performance degradation. But wavelet zerotree coding algorithms are producing variable-length codewords, extremely sensitive to bit errors. The idea is to partition the lifting coefficients. A many partition of lifting transform coefficients distributes channel error from wireless channel to each partition. Therefore synchronization problem that caused quality deterioration in still image and video stream was improved.

• Journal of Broadcast Engineering
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• v.14 no.2
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• pp.144-153
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• 2009

Recently, as the necessity of a light-weighted video encoding technique has been rising for applications such as UCC(User Created Contents) or Multiview Video, Distributed Video Coding(DVC) where a decoder, not an encoder, performs the motion estimation/compensation taking most of computational complexity has been vigorously investigated. Wyner-Ziv coding reconstructs an image by eliminating the noise on side information which is decoder-side prediction of original image using channel code. Generally the side information of Wyner-Ziv coding is generated by using frame interpolation between key frames. The channel code such as Turbo code or LDPC code which shows a performance close to the Shannon's limit is employed. The noise model of Wyner-Ziv coding for channel decoding is called Virtual Channel Noise and is generally modeled by Laplacian or Gaussian distribution. In this paper, we propose a Wyner-Ziv coding method based on the frequency-adaptive channel noise modeling in transform domain. The experimental results with various sequences prove that the proposed method makes the channel noise model more accurate compared to the conventional scheme, resulting in improvement of the rate-distortion performance by up to 0.52dB.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.43 no.1 s.343
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• pp.27-36
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• 2006

Conventional delay-insensitive (DI) data encodings usually require 2N+1 wires for transferring N-bit. To reduce complexity and power dissipation of wires in designing a large scaled chip, an encoder and a decoder circuits, where N-bit data transfer can be peformed with only N+l wires, are proposed. These circuits are based on a quasi delay-insensitive (QDI) model and designed by using current-mode multiple valued logic (CMMVL). The effectiveness of the proposed data transfer mechanism is validated by comparisons with conventional data transfer mechanisms using dual-rail and 1-of-4 encodings through simulation at the 0.25 um CMOS technology. In general, simulation results with wire lengths of 4 mm or larger show that the CMMVL scheme significantly reduces delay-power product ($D{\ast}P$) values of the dual-rail encoding with data rate of 5 MHz or more and the 1-of-4 encoding with data rate of 18 MHz or more. In addition, simulation results using the buffer-inserted dual-rail and 1-of-4 encodings for high performance with the wire length of 10 mm and 32-bit data demonstrate that the proposed CMMVL scheme reduces the D*P values of the dual-rail encoding with data rate of 4 MHz or more and 1-of-4 encoding with data rate of 25 MHz or more by up to $57.7\%\;and\;17.9\%,$ respectively.

• Journal of Broadcast Engineering
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• v.17 no.5
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• pp.742-755
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• 2012

In this paper, we propose a parallel deblocking algorithm to resolve workload imbalance when the deblocking filter of high efficiency video coding (HEVC) decoder is parallelized. In HEVC, the deblocking filter which is one of the in-loop filters conducts two-step filtering on vertical edges first and horizontal edges later. The deblocking filtering can be conducted with high-speed through data-level parallelism because there is no dependency between adjacent edges for deblocking filtering processes. However, workloads would be imbalanced among regions even though the same amount of data for each region is allocated, which causes performance loss of decoder parallelization. In this paper, we solve the problem for workload imbalance by predicting the complexity of deblocking filtering with coding unit (CU) depth information at a coding tree block (CTB) and by allocating the same amount of workload to each core. Experimental results show that the proposed method achieves average time saving (ATS) by 64.3%, compared to single core-based deblocking filtering and also achieves ATS by 6.7% on average and 13.5% on maximum, compared to the conventional uniform data-level parallelism.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.1-10
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• 2002

A space-time concatenated convolutional and differential coding scheme is employed in a multiuser direct-sequence code-division multiple-access(DS-CDMA) system. The system consists of single-user detectors (SUD), which are used to suppress multiple-access interference(MAI) with no requirement of other users' spreading codes, timing, or phase information. The space-time differential code, treated as a convolutional code of code rate 1 and memory 1, does not sacrifice the coding efficiency and has the least number of states. In addition, it brings a diversity gain through the space-time processing with a simple decoding process. The iterative process exchanges information between the differential decoder and the convolutional decoder. Numerical results show that this space-time concatenated coding scheme provides better performance and more flexibility than conventional convolutional codes in DS-CDMA systems, even in the sense of similar complexity Further study shows that the performance of this coding scheme applying to DS-CDMA systems with SUDs improves by increasing the processing gain or the number of taps of the interference suppression filter, and degrades for higher near-far interfering power or additional near-far interfering users.