• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2006.11a
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• pp.127-132
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• 2006

Quantization in video coding plays an important role in controlling the bit-rate of compressed video bit-streams. It has been used as an important control means to adjust the amount of bit-streams to at]owed bandwidth of delivery networks and storage. Due to the dependent nature of video coding, dependent quantization has been proposed and applied for MPEG-2 video coding to better maintain the quality of reconstructed frame for given constraints of target bit-rate. Since Scalable Video Coding (SVC) being currently standardized exhibits highly dependent coding nature not only between frames but also lower and higher scalability layers where the dependent quantization can be effectively applied, in this paper, we propose a dependent quantization scheme for SVC and compare its performance in visual qualities and bit-rates with the current JSVM reference software for SVC. The proposed technique exploits the frame dependences within each GOP of SVC scalability layers to formulate dependent quantization. We utilize Lagrange optimization, which is widely accepted in R-D (rate-distortion) based optimization, and construct trellis graph to find the optimal cost path in the trellis by minimizing the R-D cost. The optimal cost path in the trellis graph is the optimal set of quantization parameters (QP) for frames within a GOP. In order to reduce the complexity, we employ pruning procedure using monotonicity property in the trellis optimization and cut the frame dependency into one GOP to decrease dependency depth. The optimal Lagrange multiplier that is used for SVC is equal to H.264/AVC which is also used in the mode prediction of the JSVM reference software. The experimental result shows that the dependent quantization outperforms the current JSVM reference software encoder which actually takes a linear increasing QP in temporal scalability layers. The superiority of the dependent quantization is achieved up to 1.25 dB increment in PSNR values and 20% bits saving for the enhancement layer of SVC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.12C
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• pp.721-726
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• 2011

The compression performance of HEVC (high efficiency video coding) is improved 40%, compared to H.264/AVC. Since the existing 3D video CODEC is based on H.264/AVC or MPEG-2, we can improve the compression performance when we use the proposed stereoscopic video coding method based on HEVC. Since the stereoscopic video has the temporal and inter-view correlations, the videos of the left and right cameras encode together to improve the performance. Especially, we implemented the proposed technique using HM(HEVC test model) 3.4. To compare the performance of the proposed method, we only compare the right view video which is coded using the inter-view prediction. The proposed method which is considered inter-view correlation is improved the performance which BDBR reduce about 36.24% and BDPSNR increase approximately 1.19 dB compared to the separated-coding method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.2
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• pp.20-28
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• 2003

Recently we need a faster and more accurate motion vector search algorithm for ASIC(Application Specific IC) or small systems. Block motion estimation using Full Search(FS) algorithm provides the best visual quality and PSNR, but it requires intensive computations. The previously proposed fast algorithms reduced the number of computations by limiting the number of searching locations. This is accomplished at the expense of less accuracy of motion estimation and gives rise to an appreciably higher SAD(Sum of Absolute Difference) for motion compensated images. In this paper we exploit the spatial correlation of motion vectors and present a fast motion estimation scheme which uses the predicted motion vector(PMV). The PMV scheme is more clear and simpler than the previously proposed algorithms which also use adjacent motion vectors. Simulation results with standard video sequences show that the PMV scheme is faster and more accurate than other algorithms such as Nearest-Neighbors Search(NNS) algorithm.

• Journal of IKEEE
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• v.24 no.1
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• pp.46-51
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• 2020

With Wi-Fi and Bluetooth connectivity becoming more common in today's handheld mobile devices, single-chip multi-radio combo-modules, which integrate two or more heterogeneous wireless radios on a single chip, are becoming more and more popular. The key requirement for combo solutions is that the quality of the user experience (QoE) must not be compromised by degrading connectivity performance. Therefore, characterizing and understanding the behaviour of combo-module is of vital importance to ensure this requirement in various environments. In this paper, we investigate the impact of the use of combo-modules on the performance of mobile video streaming over a Wi-Fi network. Our study reveals that the use of combo-modules incurs considerable side effects on QoE for mobile video streaming applications when the Wi-Fi and Bluetooth operate at the same time in the 2.4GHz ISM band. We reveal that rate-based adaptive algorithms, including the most popular adaptive bitrate streaming MPEG-DASH, is more severely affected by this phenomenon than buffer-based adaptive algorithms.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.345-356
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• 2004

In this paper, we present a design and implementation of a personalized broadcasting system using TV-Anytime metadata for providing personalized services. The TV-Anytime specifies metadata schema, metadata coding and delivery, and provides service models to provide personalized broadcasting content services at anytime when users want to consume using metadata, which includes ECG (Electronic Content Guide) and content descriptive information in a PDR (Personal Digital Recorder)-centric environment. The proposed personalized broadcasting system consists of a server that provides metadata binary-coding, encapsulation and multiplexing, and a client terminal that takes charge of de-multiplexing, metadata decoding, and metadata processing for personalized content accessing and consumption. This paper presents the details of the design of each functional module, and the evaluation results with a set of service scenarios in an end-to-end broadcasting test-bed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4077-4091
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• 2017

This paper presents the design of a broadcasting scenario and system for an 8K-resolution content. Due to an 8K content is four times larger than the 4K content in terms of size, many technologies such as content acquisition, video coding, and transmission are required to deal with it. Therefore, high-quality video and audio for 8K (ultra-high definition television) service is not possible to be transmitted only using the current terrestrial broadcasting system. The proposed broadcasting system divides the 8K content into four 4K contents by area, and each area is hierarchically encoded by Scalable High-efficiency Video Coding (SHVC) into three layers: L0, L1, and L2. Every part of the 8K video content divided into areas and hierarchy is independently treated. These parts are transmitted over heterogeneous networks such as digital broadcasting and broadband networks after going through several processes of generating signal messages, encapsulation, and packetization based on MPEG media transport. We propose three methods of generating streams at the sending entity to merge the divided streams into the original content at the receiving entity. First, we design the composition information, which defines the presentation structure for displays. Second, a descriptor for content synchronization is included in the signal message. Finally, we define the rules for generating "packet_id" among the packet header fields and design the transmission scheduler to acquire the divided streams quickly. We implement the 8K broadcasting system by adapting the proposed methods and show that the 8K-resolution contents are stably received and serviced with a low delay.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.17-24
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• 2004

In this paper, the wavelet video coding using Low-Band-Shift(LBS) method and multiresolution motion estimation(MRME) is proposed. To overcome shift- variant property on wavelet coefficients, the LBS was proposed. LBS method previously has superior performance in terms of rate-distortion characteristic. However, this method needs more memory and computational complexity. Therefore to reduce computational complexity of video coding using LBS, we combine MRME with LBS. When mm is applied only, it has 7 times as much as existing method's motion vector because each subband has different motion vector using property of LBS, number of motion vector decreases. Proposed method decreases motion vector, and it decreases motion compensated Prediction error by detailed motion estimation. And then it shows better coding performance. Also this method reduces computational amount by smaller search area in higher resolution. The computational complexity of the proposed method is 12.1% of that of existing method at 3-level wavelet transform. The experimental results with the proposed method show about 0.2∼9.7% improvement of MAD performance in case of lossless coding, and 0.1∼2.0㏈ improvement of PSNR performance at 4he same bit rate in case of lossy coding.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.9
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• pp.626-636
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• 2009

In Mobile IP-enabled wireless LAN (WLAN), packet flows are corrupted due to the handoff of a mobile node (MN) at the link and network layers, which results in burst packet losses and can cause temporary buffer underflow in a streaming client at the MN. This transient behavior hurts time-sensitive streaming media applications severely. Among many suggestions to address this handoff problem, few studies are concerned with empirical issues regarding the practical validation of handoff options on the time-sensitive streaming media applications. In this paper, targeting seamless streaming over Mobile IP-enabled WLAN, we introduce a seamless media streaming framework that estimates accurate pre-buffering level to compensate the handoff latency. In addition, we propose a link-layer (L2) assisted seamless media streaming system as a preliminary version of this framework. The proposed system is designed to reduce the handoff latency and to overcome the playback disruption from an implementation viewpoint. A packet buffering and forwarding mechanism with L2 trigger is implemented to reduce the handoff latency and to eliminate burst packet losses generated during the handoff. A pre-buffering adjustment is also performed to compensate the handoff latency. The experimental results show that the proposed approach eliminates packet losses during the handoff and thus verify the feasibility of seamless media streaming over Mobile IP-enabled WLAN.

• Proceedings of the Korea Inteligent Information System Society Conference
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• 2000.04a
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• pp.307-316
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• 2000

MP3(MPEG 1 layer 3) 파일 형식(file format)은 최근 높은 압축율과 뛰어난 음질 복원 능력으로 주목을 받고 있다. 실제로 MP3의 압축율은 CD의 약 50분의 1 정도이고 음질은 CD 음질을 동일한 수준으로 유지할 수 있다.한편, 이러한 MP3의 장점 때문에 web을 통해 MP3 파일을 찾으려는 수요는 폭발적으로 증가하고 있지만 기존의 검색 엔진들이 가지고 있는 프로세스는 급속하게 update되고 있는 MP3 컨텐츠에 효과적으로 대응하지 못하고 있는 실정이다. 특히, 기존의 검색 엔진들은 미디어 파일을 위한 검색이 아닌 문자 기반의 검색 기능을 위해 개발되어 MP3 검색에는 부적절하거나, 파일 중심이 아닌 사이트 중심의 링크 변동에 대하여 수동적인 업데이트만을 수행하여 빠른 변화에 능동적으로 대응하기 어려운 경우가 많다.현재 미디어 파일을 위한 검색 엔진들은 여럿 서비스 중이지만, 텍스트 중심의 탐색 방법을 사용하고, 정기적인 DB update 방법에 관해서도 문자 기반의 검색 엔진과 동일한 방법을 사용하고 있다. 또한, 국내에서는 web 서비스를 위한 미디어 파일 탐색 알고리즘과 지능형 탐색 방법에 등에 관한 연구 역시 거의 전무한 상태이다.본 논문은 MP3 파일 전문 검색을 위한 지능형 프로세스를 설계와 구현 결과에 관한 것으로, 기존의 미디어 검색 엔진들이 가지는 문제점을 지적하고 보다 효율적이고 능동적인 미디어 파일 탐색을 위한 방법을 제시한다. 특히, MP3 파일에 대한 미디어 파일 검증 알고리즘과 verification method을 제안하고, 이러한 메커니즘에 따라 구현된 지능형 robot과 spider 등으로 구성된, 신뢰성 있고 지능적인 MP3 검색 엔진 지원 시스템의 설계와 구현 결과 그리고 성능 등을 종합적으로 요약한다.실어증 환자들은 화시적 대명사를 조응적 대명사보다 더 잘 처리하는 동일한 결과를 보였다. 이러한 실험 결과들은 실어증 환자들이 뇌손상으로 인해 문법적 언어처리에는 어려움을 보이지만 비언어적인, 세상 지식과 관련된 화시적 대명사의 처리는 가능할 것이라는 가설을 뒷받침 해준다. 또한 이러한 실험 결과를 통해 대명사의 기능적인 측면에서 화시와 조응의 처리가 구분되어 있음을 보여준다.l mechanism is concentrate on only the reaction zone. As strain rate and CO2 quantity increase, NO production is remarkably augmented.our 10%를 대용한 것이 무첨가한 것보다 많이 단단해졌음을 알 수 있었다. 혼합중의 반죽의 조사형 전자현미경 관찰로 amarans flour로 대체한 gluten이 단단해졌음을 알수 있었다. 유화제 stearly 칼슘, 혹은 hemicellulase를 amarans 10% 대체한 밀가루에 첨가하면 확연히 비용적을 증대시킬 수 있다는 사실을 알 수 있었다. quinoa는 명아주과 Chenopodium에 속하고 페루, 볼리비아 등의 고산지에서 재배 되어지는 것을 시료로 사용하였다. quinoa 분말은 중량의 5-20%을 quinoa를 대체하고 더욱이 분말중량에 대하여 0-200ppm의 lipase를 lipid(밀가루의 2-3배)에 대하여 품질개량제로서 이용했다. 그 결과 quinoa 대량 7.5%에서 비용적, gas cell이 가장 긍정적 결과를 산출했고 반죽의 조직구조가 강화되었다. 또 quinoa 대체에 의해 전분-지질 복합제의 흡열량이 증대된 것으로부터 전분-지질복합제의 형성 촉진이 시사되었다.이것으로 인하여 호화억제에 의한 노화 방지효과가 기대되었지만