• 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.52 no.5
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• pp.3-9
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• 2015

The wireless communication system adopts an appropriate retransmission scheme on each system protocol layer to improve reliability of data transmission. In each system protocol layer, the retransmission scheme operates in independently other layers and operates based on the parameters without reference to end-to-end performance of wireless communication system. For this reason, it is difficult to design the optimal system parameters that satisfy the QoS requirements for each service class. Thus, the performance analysis of wireless communication system is needed to design the optimal system parameters according to the end-to-end QoS requirements for each service class. In this paper, we derive the mathematical model to formulate the end-to-end performance of wireless communication system. We also evaluate the performance at the MAC and transport layers in terms of average spectral efficiency and average transmission delay. Based on the results of performance evaluations, we design the optimal system parameters according to the QoS requirements of service classes. From the results, the HARQ combined with AMC is appropriate for the delay-sensitive service and the ARQ combined with AMC is appropriate for a service that is insensitive to transmission delay. Also, the TCP can be applied for the delay-insensitive service only.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.5
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• pp.11-20
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• 2002

Based on regularization principles, we propose a new dequantization scheme on DCT-based transform coding for reducing of blocking artifacts and minimizing the quantization error. The conventional image dequantization is simply to multiply the received quantized DCT coefficients by the quantization matrix. Therefore, for each DCT coefficients, we premise that the quantization noise is as large as half quantizer step size (in DCT domain). Our approach is based on basic constraint that quantization error is bounded to ${\pm}$(quantizer spacing/2) and at least there are not high frequency components corresponding to discontinuities across block boundaries of the images. Through regularization, our proposed dequantization scheme, sharply reduces blocking artifacts in decoded images. Our proposed algorithm guarantees that the dequantization process will map the quantized DCT coefficients will be evaluated against the standard JPEG, MPEG-1 and H.263 (with Annex J deblocking filter) decoding process. The experimental results will show visual improvements as well as numerical improvements in terms of the peak-signal-to-noise ratio (PSNR) and the blockiness measure (BM) to be defined.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.307-314
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• 2003

In the general multiple video object coder, more interested objects such as speaker or moving object is consistently coded with higher priority. Since the priority of each object may not be fixed in the whole sequence and be variable on frame basis, it must be adjusted in a frame. In this paper, we analyze the independent rate control algorithm and global algorithm that the QP value is controled by the static parameters, object importance or priority, target PSNR, weighted distortion. The priority among static parameters is analyzed and adjusted into dynamic parameters according to the visual interests or importance obtained by camera interface. Target PSNR and weighted distortion are proportionally derived by using magnitude, motion, and distortion. We apply those parameters for the weighted distortion control and the priority-based control resulting in the efficient bit-rate distribution. As results of this paper, we achieved that fewer bits are allocated for video objects which has less importance and more bits for those which has higher visual importance. The duration of stability in the visual quality is reduced to less than 15 frames of the coded sequence. In the aspect of PSNR, the proposed scheme shows higher quality of more than 2d13 against the conventional schemes. Thus the coding scheme interfaced to human- eye proves an efficient video coder dealing with the multiple number of video objects.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4A
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• pp.451-463
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• 2000

In this paper, we design the radio data link protocols with QoS provisioning for mobile multimedia such as voice, data, and video in CDMA-based ATM networks, and analyze the performance of the data link protocols. To support mobile multimedia traffic, the required QoS parameters and the characteristics are analyzed, and wireless protocol stacks are proposed for integrating the wireless access network and ATM transport networks, and radio data link protocols are designed for provisioning QoS Control. The data link protocols are analyzed assuming that the system is supporting voice and data traffic simultaneously. In case of data traffic, the delay and throughput of SREJ ARQ and Type-1 Hybrid ARQ scheme are compared, and in case of voice traffic, the packet loss rate of BCH coding is analyzed according to the varying data traffic loads. The results indicate that the adaptive radio link protocols are efficient to support QoS requirements while the complexities are increased.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8A
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• pp.879-888
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• 2004

In this paper, we propose a real time QoS(Quality of Service) guaranteed transmission mechanism for MPEG video traffic at the congested node in IP based networks. Recent spread of the Internet has increased the demands of a real time multimedia service of the quality, Because the type of Internet services can, however, offer the best effort delivery strategies, it is difficult to treat all the types of traffic with differential COS (Class of Service). Most of all, the hierarchical coding method of MPEG data utilizes the reference frame for the motion prediction. The loss of the reference frames makes QoS of the video traffic degraded because the reference frame bit error causes the consecutive packet loss. Therefore we have studied the real time QoS guaranteed mechanism for video traffic by analyzing the previous methods. Computer simulation results show that the proposed scheme has better performance than the previous one.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.3
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• pp.254-262
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• 2013

This paper proposes FMT(Filtered Multi-Tone)-based digital radio implementation algorithms and the results obtained by various field tests especially in terms of transmitter characteristics. In this study, we predefined frame structure and protocols used for the CSTDMA(Carrier Sensing Time Division Multiple Access) scheme, designed digital filters and RF front end to fulfill the system characteristics such as the spectral mask and processing delays given by the Recommendation ITU-R M.1842-1. The proposed system supports exchange of data for e-Navigation with the usage of wider channel of 50-100kHz bandwidth, Turbo coding and FMT modulation. Furthermore, the common Ethernet protocol makes connection to local WLAN(Wireless Local Area Network) on board the ship for other data services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12C
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• pp.965-970
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• 2010

In the NAND flash memory storing two bits per cell, data is discriminated among four levels of electrical charges. We refer to these four levels as E, P1, P2, and P3 from the low voltage. In the statistics, many errors occur when E and P3 are stored at the next cells. Therefore, we propose a coding scheme for avoiding E-P3 or P3-E data patterns. We investigate two modulation codes for 9/10 code (9 bit input and 5 symbol codeword) and 11/12 code (11 bit input and 6 symbol codeword).

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.45 no.11
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• pp.77-82
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• 2008

H.264 is a video coding standard of ITU-T and ISO/IEC, and widely spreads its application due to its high compression ratio more than twice that of MPEG-2 and high image quality. It has different architecture depending on demands since it is a lied from small image of QVGA to large size of HD. In this paper, We propose a scalable architecture for intra-prediction of H.264 decoders. The proposed scheme has a scalable architecture that can accommodate up to 4 processing elements depending on performance demands and can reduce the number of access to memory using efficient memory management so as to be energy-efficient. We design the intra-prediction unit using Verilog-HDL and verily it by prototyping using an FPGA. The performance is analyzed using the results of design.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.9 no.2
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• pp.60-67
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• 2010

In the ubiquitous age, applications of wireless personal area network (WPAN) technology using LEDs are in progress. However, visible light communications (VLC) using the LEDs have weakness which deteriorate performance of communication because of multi-path fading that occurs propagation delay by interior walls or other things in indoor environments. In this paper, orthogonal frequency division multiplexing (OFDM) scheme is adapted to decrease multi-path fading and multi-path dispersion and to provide high speed data transmission. Besides, to reduce information losses caused by optical noise (incandescent lamps, fluorescent lamps, sunbeam etc.) also proposed channel coding using turbo codes. The encoding and decoding of the proposed system is described, and simulation results are analyzed. We can know that performance of proposed system is increased about 4 [dB] through the simulation results. Also, when the system take doppler effect, the system performance worsened.