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Next Generation Mobile Video Networking

ZTE communications 2018年第3期16卷 1-2页

作者： hwang jenq-neng wen yonggang Global Affairs and Inter⁃national Development in the EE Department Multimedia Techni⁃cal Committee(MMTC)of IEEE Communication Society Multimedia Signal Processing Technical Committee(MMSP TC)of IEEE Signal Processing Society IEEE T⁃SP T⁃NN andT⁃CSVTT⁃IP andSignal Processing Magazine(SPM) ZTE Communications ETRIIJDMB andJSPS journals the School of Computer Science and Engineering Nanyang Technological UniversitySingapore

HWANG Jenq-Neng received his Ph.D. degree from the University of Southern California, USA. In the summer of 1989, Dr. HWANG joined the De- partment of Electrical Engineering of the Universi- ty of Washington in Seattle, USA, where he has been promoted to Full Professor since 1999. He served as the Associate Chair for Research fi＇om 2003 to 2005, and from 2011-2015. He is current- ly the Associate Chair for Global Affairs and Inter- national Development in the EE Depamnent. Hehas written more than 330 journal papers, conference papers and book chapters in the areas of machine learning, muhimedia signal processing, and muhimedia system integration and networking, including an au- thored textbook on ＂Multimedia Networking： from Theory to Practice,＂ published by Cambridge University Press. Dr. HWANG has close work- ing relationship with the industry on muhimedia signal processing and nmltimedia networking.

关键词： QoE DAF Next Generation Mobile Video Networking

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Platform-based MPEG-4 SOC design for video communications

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Journal of VLSI signal processing Systems for signal, Image, and Video Technology 2006年第1期42卷 7-19页

作者： Chang, Yung-Chi Chao, Wei-Min Hsu, Chih-Wei Chen, Liang-Gee Department of Electrical Engineering Graduate Institute of Electronics Engineering National Taiwan University Taipei Taiwan Department of Electrical Engineering GIEE NTU Taipei Taiwan Electronics Research and Service Organization Industrial Technology Research Institute Hsinchu Taiwan IEEE CAS Technical Committee of VLSI Systems and Applications Technical Committee of Visual Signal Processing and Communications IEEE Signal Processing Technical Committee of Design and Implementation of SP Systems IEEE CAS Technical Committee on Multimedia Systems and Applications Phi Tan Phi

An MPEG-4 video coding SOC design is presented in this paper. We adopt platform-based architecture with an embedded RISC core and efficient memory organization. A motion estimator supporting predictive diamond search and spiral full search is implemented for compromise between compression performance and design cost. The proposed data reuse scheme reduces required memory access bandwidth. For texture coding path, an interleaving DCT/IDCT scheduling with substructure sharing technique is proposed. Several key modules are integrated into an efficient platform in hardware/software co-design fashion. The cost-efficient video encoder SOC consumes 256.8 mW at 40 MHz and achieves real-time encoding of 30 CIF (352×288) frames per second. © Springer Science + Business Media, Inc. 2006.

关键词： Multimedia systems

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Journal of Information Science and Engineering: Foreword

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Journal of Information Science and Engineering 2006年第5期22卷

作者： Wang, Jhing-Fa Chung, Pau-Choo Billinghurst, Mark National Cheng Kung University Tainan Taiwan Center for Research of E-life Digital Technology National Cheng Kung University Taiwan Electrical Laboratory National Cheng Kung University Taiwan Technical Committee of Bioinformatics and Bioengineering IEEE Life Science Systems and Applications Technical Committee IEEE Visual Signal Processing Communication Technical Committee IEEE Neural Systems and Applications Technical Committee Multimedia Systems and Applications Technical Committee in the CASS United States IEEE Phi Tau Phi Honor Society United States IEEE International Steering Committee Taiwan University of Canterbury New Zealand

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Statistical multiplexing based on MPEG-4 fine granularity scalability coding

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Journal of VLSI signal processing Systems for signal, Image, and Video Technology 2006年第1期42卷 69-77页

作者： Yang, Xiaokang Ling, Nam Institute of Image Communication and Information Processing Shanghai Jiao Tong University 800 Dong Chuan Road Shanghai 200240 China Department of Computer Engineering Santa Clara University 500 El Camino Real Santa Clara CA 95053 United States Institute of Image Communication and Information Processing Department of Electronic Engineering Shanghai Jiao Tong University Shanghai China Visual Signal Processing Communications Technical Committee IEEE Circuits and Systems Society Department of Computer Engineering California United States National University of Singapore IEEE SPIE ASEE

Conventional statistical multiplexing methods for MPEG-1/2 programs involve high computationally complex transcoding (decoding and re-encoding) process to convert the original bit-rate into the target bit-rate. To avoid a time-consuming transcoding process, a new statistical multiplexer is proposed to enable content provider to easily convert the bit-rates by exploiting the MPEG-4 fine granularity scalability (FGS) coding scheme. The proposed statistical multiplexer is particularly useful for multiple-program broadcasting applications, including Internet television and video on demand, as well as value-added MPEG-4 video streaming services for DVB and ATSC digital TV systems. The proposed multiplexer mainly includes two parts: the FGS-based frame lag scheme and the optimal bit-plane truncation scheme. The FGS-based frame lag scheme exploits intra- and inter-layer correlations exist in MPEG-4 FGS bit-streams. The optimal bit-plane truncation scheme dynamically truncates enhancement layers of FGS bit-streams under the available bandwidth constraint and the quality/smoothness constraint. Experimental results show that high statistical multiplexing efficiency, inter-program fairness, and intra-program smoothness are achieved by the proposed multiplexer. © Springer Science + Business Media, Inc. 2006.

关键词： Multimedia systems

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A new efficient block-matching algorithm for motion estimation

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Journal of VLSI signal processing Systems for signal, Image, and Video Technology 2006年第1期42卷 21-33页

作者： Mahmoud, Hanan Goel, Sumeer Shaaban, Mohsen Bayoumi, Magdy Faculty of Engineering University of Alexandria University Buildings 9 Albert Street Semouha Alexandria 21321 Egypt Center for Advanced Computer Studies University of Louisiana at Lafayette 110 Rex Street Lafayette LA 70504 Faculty of Engineering University of Alexandria Computer Science Department Department of Computer Engineering Department of Computer Science Center for Advanced Computer Studies University of Louisiana Lafayette Technical Committee on Circuits and Systems for Communication Technical Committee on Signal Processing Design and Implementation Neural Network and the Multimedia Technology Technical Committees University of Louisiana Lafayette

This paper presents a new full-search block-matching algorithm: Multi-stage Interval-based Motion Estimation algorithm (MIME). The proposed algorithm is a block based motion estimation algorithm that utilizes successive elimination technique. We define two approximate functions, as the upper and lower boundaries of the interval that includes the Conventional distortion metric SAD. Each stage in the proposed algorithm;except for the last stage;incorporates low resolution pixels for the boundary functions calculations. The final stage is a full resolution block matching stage. MIME has a high probability of finding the optimal motion vector at any stage of the algorithm. The proposed algorithm reduces the computational complexity by successively eliminating non-candidate blocks from the search window at each stage. This computational reduction leads to enhanced performance in terms of low power consumption and fast motion vector estimation. A low power VLSI implementation of the algorithm is also presented in this paper. Simulation results on benchmark video sequences shows that MIME algorithm eliminates almost 88% of the candidate blocks after only two interval based stages. © Springer Science + Business Media, Inc. 2006.

关键词： Motion estimation

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Using hybrid HMM/BN acoustic models: Design and implementation issues

Using hybrid HMM/BN acoustic models: Design and implementati...

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作者： Markov, Konstantin Nakamura, Satoshi ATR Spoken Language Communication Research Labs. Kyoto-fu 619-0288 Japan Acoustics and Speech Processing Department ATR SLC Japan Acoustical Society of Japan Japan IEEE Japan ISCA Japan Acoustics and Speech Research Department ATR Spoken Language Translation Laboratories Japan Speech Technical Committee IEEE Signal Processing Society United States Information Processing Society of Japan Japan

In recent years, the number of studies investigating new directions in speech modeling that goes beyond the conventional HMM has increased considerably. One promising approach is to use Bayesian Networks (BN) as speech models. Full recognition systems based on Dynamic BN as well as acoustic models using BN have been proposed lately. Our group at ATR has been developing a hybrid HMM/BN model, which is an HMM where the state probability distribution is modeled by a BN, instead of commonly used mixtures of Gaussian functions. In this paper, we describe how to use the hybrid HMM/BN acoustic models, especially emphasizing some design and implementation issues. The most essential part of HMM/BN model building is the choice of the state BN topology. As it is manually chosen, there are some factors that should be considered in this process. They include, but are not limited to, the type of data, the task and the available additional information. When context-dependent models are used, the state-level structure can be obtained by traditional methods. The HMM/BN parameter learning is based on the Viterbi training paradigm and consists of two alternating steps - BN training and HMM transition updates. For recognition, in some cases, BN inference is computationally equivalent to a mixture of Gaussians, which allows HMM/BN model to be used in existing decoders without any modification. We present two examples of HMM/BN model applications in speech recognition systems. Evaluations under various conditions and for different tasks showed that the HMM/BN model gives consistently better performance than the conventional HMM. Copyright © 2006 The Institute of Electronics, Information and communication Engineers.

关键词： Speech recognition

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From rate-distortion analysis to resourcedistortion analysis

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IEEE Circuits and Systems Magazine 2005年第3期5卷 6-18页

作者： He, Zhihai Mitra, Sanjit K. Department of Electrical and Computer Engineering University of Missouri Columbia MO 65211 Department of Electrical and Computer Engineering University of California Santa Barbara CA Visual Signal Processing and Communication Technical Committee IEEE Circuits and Systems Society Academy of Finland Croatian Academy of Sciences and Arts IEEE AAAS SPIE EURASIP ASEE

The ultimate goal in communication system design is to control and optimize the system performance under resource constraints. As the communication paradigm evolves from the conventional desktop computing, wired, and centralized communication to current mobile, wireless, distributed, and massive communication, video encoding and transmission operate under more and more resource constraints. In traditional video communication applications, such as digital TV broadcast, the major constraint is in the form of transmission bandwidth or storage space, which determines the encoding bit rate. Ratedistortion (R-D) theories have been developed to model the relationship between the coding bit rate and signai distortion. For video communication over mobile devices, the video encoding and transmission operate under additional resource constraints, such as energy supply and on-board computation capability. Therefore, there is a need to extend the traditional R-D analysis to resource-distortion analysis by incorporating the new resource constraints into the R-D analysis framework. In distributed and massive wireless video sensor networks, the resource utilization behaviors of individual video sensors should be well-coordinated through network-level rate allocation and optimum routing so as to maximize the overall performance. In this paper, we start from the classical R-D theory developed by Shannon over 50 years ago, and then review the R-D modelling techniques for modern image and video compression systems. We study the resource-distortion analysis framework for video communication over wireless devices. As one step further, we present the research problem of resource allocation and performance optimization for video compression and communication over a network of wireless communication devices. © 2005 IEEE.

关键词： communication systems

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