• The KIPS Transactions:PartB
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• v.17B no.1
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• pp.31-36
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• 2010

After the launch of MPEG-2, it is widely used in multimedia applications like a Digital-TV or a DVD. Then, After the launch of H.264 at 2004, it has been expected to replace MPEG-2 and services IPTV and DMB. As we have been used to MPEG-2 devices by this time, we can not access H.264 Broadcast with MPEG-2 device. So We propose a new approach to transcode H.264 video into MPEG-2 form which can facilitate to display H.264 video with MPEG-2 device. To reduce the quality loss by transcoding, we use CPDT(Cascaded Pixel Domain Transcoder) structure. And to minimize processing time, SKIP block, INTRA block and motion vectors obtain from decoding process is employed for transcoding. we use BMA(Boundary Matching Algorithm) to select only one from candidate motion vectors. Experimental results show a considerable improved PSNR with reduction in processing time compared with existing methods.

• Journal of Advanced Marine Engineering and Technology
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• v.40 no.1
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• pp.45-49
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• 2016

The performance of underwater acoustic communication systems is sensitive to inter-symbol interference caused by delay spread developed from multipath signal propagation. The multipath nature of underwater channels causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Of the various iterative coding schemes, turbo code and convolutional code based on the BCJR algorithm have recently dominated this application. In this study, the performance of iterative codes based on turbo equalizers with equivalent coding rates and similar code word lengths were analyzed. Underwater acoustic communication system experiments using these two coding techniques were conducted on Kyeong-chun Lake in Munkyeong City. The distance between the transmitter and receiver was 400 m, and the data transfer rate was 1 Kbps. The experimental results revealed that the performance of turbo codes is better for channeling than that of convolutional codes that use a BCJR decoding algorithm.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.11
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• pp.24-29
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• 2010

Selection-type relaying protocol, which is one of cooperative relaying protocols, provides low decoding complexity and improved system performance due to selection diversity. In this paper, we deal with linear precoding technique that minimize the error probability of cooperative MIMO system. Under the assumption that full channel state information is available at whole nodes, linear source and relay precoders, which minimize mean squared error of the estimated symbol vector, are proposed. Moreover, unlikely to the conventional selection-type relaying protocol using a fixed threshold signal-to-noise-ratio, new transmission link selection algorithm selects direct link or relay link as a transmission link, is introduced. Simulation results show that the proposed linear precoder with the transmission link selection algorithm outperforms the conventional precoders for two-hop relaying protocols or selection-type relaying protocols.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1393-1399
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• 1999

In this paper, a VLSI design of the convolutional codec chip of code rate r=l/3, and constraint length K=9 is presented, which is able to correct errors of the received data when transmitted data is corrupted in channels. The circuit design mainly aimed for simple implementation. In the decoder, Viterbi algorithm with 3-bit soft-decision is employed. For information sequence updating and storage, the register exchange method is employed, where the register length is 5$\times$K(45 stages). The codec chip is designed using VHDL language and Design Analyzer and VHDL Simulator of Synopsys are used for simulation and synthesis. The chip is composed of ENCODER block, ALIGN block, BMC block, ACS block, SEL_MIN block and REG_EXCH block. The operation of the codec chip is verified though the logic simulations, where several error conditions are assumed. As a result of the timing simulation after synthesis, the decoding speed of 325.5Kbps is achieved, and 6,894 gates is used.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3A
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• pp.217-225
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• 2005

This paper presents specialized DSP instructions and their architecture for the Viterbi algorithm used in various wireless communication standards. The proposed architecture can significantly reduce the Trace Back (TB) latency. The proposed instructions perform the Add Compare Select (ACS) and TB operations in parallel and the architecture has special hardware, called the Offset Calculation Unit (OCU), which automatically calculates data addresses for the trellis butterfly computations. Logic synthesis has been Performed using the Samsung SEC 0.18 μm standard cell library. OCU consists of 1,460 gates and the maximum delay of OCU is about 5.75 ns. The BER performance of the ACS-TB parallel method increases about 0.00022dB at 6dB Eb/No compared with the typical TB method, which is negligible. When the constraint length K is 5, the proposed DSP architecture can reduce the decoding cycles about 17% compared with the Carmel DSP and about 45% compared with 7MS320c15x.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3C
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• pp.147-152
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• 2005

In this paper, we proposed a method for compressing the static video test patterns in high quality to test the picture quality in DTV. In our method, we use the fact that the generated bits and average quantization value have almost identical distribution characteristics per each GOP and we propose a new target bit allocation method suitable for compressing the static test pattern while the target bit allocation method in MPEG-2 TM5 is suitable for the moving picture. The proposed target bit allocation method is to maintain the high quality video continuously by using the normalized complexities which are updated or maintained by means of picture qualities at each GOP. Experiment result showed that the test pattern stream encoded by MPEG-2 software with the proposed algorithm had a stable bit rate and good video quality during the decoding process.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.1-10
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• 2004

This study discusses Application-layer FEC using erasure codes. Because of the simple decoding process, erasure codes are used effectively in Application-layer FEC to deal with Packet-level errors. The large number of parity packets makes the loss rate to be small, but causes the network congestion to be worse. Thus, a redundancy control algorithm that can adjust the number of parity packets depending on network conditions is necessary. In addition, it is natural that high-priority frames such as I frames should produce more parity packets than low-priority frames such as P and B frames. In this paper, we propose a redundancy control algorithm that can adjust the amount of redundancy depending on the network conditions and depending on data priority, and test the performance in simple links and congestion links.

• Journal of Broadcast Engineering
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• v.14 no.4
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• pp.387-396
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• 2009

No-Reference video-quality assessments are divided into two kinds of metrics based on decoding pixel domain or the bitstream one. Traditional full-/reduced- reference methods have difficulty to be deployed as realtime video transmission because it has problems of additional data, complexity, and assessment accuracy. This paper presents simple and highly accurate no-reference video-quality assessment in realtime video transmission. Our proposed method uses quantization parameter, motion vector, and information of transmission error. To evaluate performance of the proposed algorithm, we perform subjective test of video quality with the ITU-T P.910 Absolute Category Rating(ACR) method and compare our proposed algorithm with the subjective quality assessment method. Experimental results show the proposed quality metric has a high correlation (85%) in terms of subjective quality assessment.

• Journal of Broadcast Engineering
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• v.4 no.1
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• pp.76-85
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• 1999

The degradation of image quality by block loss is more serious in fractal coded images with the error propagation due to mapping from the lost blocks than in DCT coded images. Therefore. a new algorithm is presented for recovering the blocks lost in the transmission through the lossy network as A TM network of the images coded by Jacquins fractal coding. Jacquins fractal code is divided into two layers of header code and main code according to its importance. The key technique of the proposed BLRA (block loss recovery algorithm) is a fractal extrapolation that estimates the lost pixels by using the contractive mapping parameters of the neighboring range blocks whose characteristics are similar to a lost block. The proposed BLRA is applied to the lost blocks in the iteration of decoding. Some experimental results show the proposed BLRA yields excellent performance in PSNR as well as subjective quality.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.42-49
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• 1989

A new integrated coding method is proposed in this study for embedding the text information including Hangul into an image. A monochrome analog image may be quantized to a few leveled digital image and be displayed on bi-leveled output devices by using halftone processing techniques. Text data are embedded on each micro pattern. Based on this concept, the encoding and the decoding algorithm are implemented and experiments are performed. As a result, the average amount of the embedded text information is more than 8 bpp (bits per pixer) in this halftone processed image converted form a $64{\times}64$ image, i.e, corresponding to 2000 characters in Hangul, or 4000 characters in alphanumeral. using this algorithm, the integrated personal record management system is implemented.