Browsing by Author "Liu, JC"

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A MAC protocol for real-time manufacturing traffic in client-server-based high speed broadcast networks
(1996)
Liu, JC
;
Yuang, MC
;
Wu, CHJ
;
交大名義發表
;
資訊工程學系
Advances in network technology and the emergence of applications for manufacturing demanding stringent Quality of Service (QoS) requirements have made the design of high-speed broadcast networks feasible and indispensable. Unlike traditional broadcast networks, such as Ethernet, high-speed broadcast networks exhibit high bandwidth-delay product. Moreover, forthcoming networks are required to support diverse virtual manufacturing devices demanding different amounts of bandwidth. Examples are networks with client-server configurations. These facts render traditional Medium Access Control (MAC) protocols unviable. The paper proposes a new MAC protocol, called Prioritized Adaptive Cycle Cell Insertion (PACCI), especially advantageous to client-server-based high-speed broadcast networks. PACCI achieves bounded delay and high utilization regardless of the network size and load. In addition to providing fair access for clients through c_cycles, PACCI offers prioritized access to servers through s_cycles. The bandwidth allocation of the c_cycles and s_cycles is then based on an analytic model in an effort to provide QoS guarantees in terms of throughput under diverse traffic loads. The paper also presents simulation results that demonstrate the accuracy of the analysis and the superior performance of PACCI.
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-08-01)
Liu, CM
;
Liu, JC
;
交大名義發表
;
資訊工程學系
;
National Chiao Tung University
In MPEG I audio compression standard, the intensity stereo coding scheme is a keg, technology to achieve a bit rate lower than 2x128 kbit/s for stereophonic signals while preserving high audio quality. However, the intensity coding scheme has difficulties in signal cancellation and scalefactor consistency. This paper shows that the Karhunen-Loeve (KL) transform can be applied to the intensity coding for solving the two problems. In this paper, we present true methods to implement the Karhunen-Loeve transform in the MPEG 1 layers I and II. Among the two methods, one is fully compatible with the MPEG1 while the other need a slight coding of modification on the MPEG coding process. All the conclusion. two methods are tested to have better audio quality than the technique indicated by the MPEG 1 layers 1 and II.
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-01-01)
Liu, JC
;
Liu, CM
;
資訊工程學系
;
Department of Computer Science
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-08-01)
Liu, CM
;
Liu, JC
;
資訊工程學系
;
Department of Computer Science
In MPEG I audio compression standard, the intensity stereo coding scheme is a keg, technology to achieve a bit rate lower than 2x128 kbit/s for stereophonic signals while preserving high audio quality. However, the intensity coding scheme has difficulties in signal cancellation and scalefactor consistency. This paper shows that the Karhunen-Loeve (KL) transform can be applied to the intensity coding for solving the two problems. In this paper, we present true methods to implement the Karhunen-Loeve transform in the MPEG 1 layers I and II. Among the two methods, one is fully compatible with the MPEG1 while the other need a slight coding of modification on the MPEG coding process. All the conclusion. two methods are tested to have better audio quality than the technique indicated by the MPEG 1 layers 1 and II.
A tool portfolio elimination mechanism (TPEM) for a wafer fab
(2004)
Chung, SH
;
Hsieh, MH
;
Liu, JC
;
工業工程與管理學系
;
Department of Industrial Engineering and Management
Wafer manufacturers usually face critical problems when making decisions regarding to the tool portfolio elimination, mainly due to the dramatic and frequent influences from their internal and external environments. This paper is aimed to develop a mechanism, named Tool Portfolio Elimination Mechanism (TPEM), to evaluate the impacts on production performance and capital expenditure, and to determine which equipment is suitable for pruning. In TPEM, there are four stages to decide tool portfolio elimination according to the PDCA cycle. Meanwhile, the simulation results show; that the fab production performance and cost are much improved by the TPEM algorithm. Especially, it can be applied and implemented to industry very quickly and easily.
DMTS: A distributed multimedia teleworking system
(1998-11-01)
Yuang, MC
;
Liu, JG
;
Chen, YG
;
Liu, JC
;
資訊工程學系
Multimedia systems combine a variety of information sources, such as voice, graphics, animation, images, and full-motion video, into a wide range of applications. The paper initially categorizes existing multimedia applications into three classes: noninteractive-oriented, interactive-oriented client-server-based, and interactive-oriented peer-party-based. In particular, the paper examines interactive-oriented applications and provides an in-depth survey of the media synchronization problem for the design of these applications. The paper then presents our prototyping system, called the Distributed Multimedia Teleworking System (DMTS), which allows two or more remote systems in collaboration to access and modify multimedia data through a network in a fully synchronous fashion. The system has been developed over TCP/IP and an FDDI network, using an XVideo D/A card. The media supported by DMTS include text, graphics, voice, and video. DMTS employs a master-slave collaboration model to maintain the coherence of the text and graphics data being simultaneously modified. Moreover, DMTS also adopts effective mechanisms to reduce skew (asynchrony) and jitter delays between video and voice streams. Finally, the paper demonstrates that DMTS achieves a maximum throughput of 13 frames per second, and reveals that the throughput bottleneck resides in the hardware capture and D/A processing of video frames.
DMTS: A distributed multimedia teleworking system
(1998-11-01)
Yuang, MC
;
Liu, JG
;
Chen, YG
;
Liu, JC
;
資訊工程學系
Multimedia systems combine a variety of information sources, such as voice, graphics, animation, images, and full-motion video, into a wide range of applications. The paper initially categorizes existing multimedia applications into three classes: noninteractive-oriented, interactive-oriented client-server-based, and interactive-oriented peer-party-based. In particular, the paper examines interactive-oriented applications and provides an in-depth survey of the media synchronization problem for the design of these applications. The paper then presents our prototyping system, called the Distributed Multimedia Teleworking System (DMTS), which allows two or more remote systems in collaboration to access and modify multimedia data through a network in a fully synchronous fashion. The system has been developed over TCP/IP and an FDDI network, using an XVideo D/A card. The media supported by DMTS include text, graphics, voice, and video. DMTS employs a master-slave collaboration model to maintain the coherence of the text and graphics data being simultaneously modified. Moreover, DMTS also adopts effective mechanisms to reduce skew (asynchrony) and jitter delays between video and voice streams. Finally, the paper demonstrates that DMTS achieves a maximum throughput of 13 frames per second, and reveals that the throughput bottleneck resides in the hardware capture and D/A processing of video frames.
PACCI: A high performance MAC protocol for client-server-based gigabit LANs and MANs
(1996-05-01)
Yuang, MC
;
Liu, JC
;
交大名義發表
;
National Chiao Tung University
QTS: A QOS-guaranteed transport system for broad-band multimedia communications
(1998-02-01)
Yuang, MC
;
Liu, JC
;
資訊工程學系
;
Department of Computer Science
In this paper, we propose a quality of-service (QOS)guaranteed transport system (QTS), which offers various QOS's at the transport layer for broad-band multimedia communications, The QTS, composed of a bandwidth allocator and a transport protocol module, supports three classes of applications requiring different bit rates, delay sensitivity, and loss sensitivity, The bandwidth allocator intelligently manages the allocation of transport-layer bandwidth at the expense of imposing inevitable blocking of delay-sensitive application connections, The transport protocol module of the QTS performs rate-based flow control for delay-sensitive applications based on transfer rates predetermined by the bandwidth allocator, In addition, the module accomplishes error control only for loss-sensitive applications, As a result, as will be shown, by providing guaranteed rates and reducing the error control overhead, the QTS offers satisfactory bounded delays and jitters for delay-sensitive applications, while incurring minimal throughput degradation for loss-sensitive applications, Finally, we demonstrate the superiority of the QTS over transmission control protocol (TCP) via simulation results in terms of maximum and mean system delays and delay jitter.
QTS: A QOS-guaranteed transport system for broad-band multimedia communications
(1998-02-01)
Yuang, MC
;
Liu, JC
;
資訊工程學系
;
Department of Computer Science
In this paper, we propose a quality of-service (QOS)guaranteed transport system (QTS), which offers various QOS's at the transport layer for broad-band multimedia communications, The QTS, composed of a bandwidth allocator and a transport protocol module, supports three classes of applications requiring different bit rates, delay sensitivity, and loss sensitivity, The bandwidth allocator intelligently manages the allocation of transport-layer bandwidth at the expense of imposing inevitable blocking of delay-sensitive application connections, The transport protocol module of the QTS performs rate-based flow control for delay-sensitive applications based on transfer rates predetermined by the bandwidth allocator, In addition, the module accomplishes error control only for loss-sensitive applications, As a result, as will be shown, by providing guaranteed rates and reducing the error control overhead, the QTS offers satisfactory bounded delays and jitters for delay-sensitive applications, while incurring minimal throughput degradation for loss-sensitive applications, Finally, we demonstrate the superiority of the QTS over transmission control protocol (TCP) via simulation results in terms of maximum and mean system delays and delay jitter.