This letter presents two high-performance level-converting flip-flops (LCFF) for multi-VDD systems, indirect precharging flip-flop (IPFF) and multi-supply complementary pass-transistor flip-flop (MCPFF). Employing a simple precharging scheme, IPFF provides high operating speed. MCPFF, on the other hand, provides low power operations by implementing the edge-triggering function with complementary pass transistors. Performance comparison indicates that IPFF operates at the highest speed and MCPFF consumes the lowest power among the seven LCFFs under evaluation.

According to the Gestalt principals, this paper presents a recognition method of grouping areas in trademark images modeling features for measuring the attraction degree between couples of image components. This investigation would be used for content-based image retrieval from the view of mirroring human perception for images. Depending on variability in human perception for trademark images, the proposed method finds grouping areas by calculating Mahalanobis distance with the features to every combination of two components in images. The features are extracted from every combination of two components in images, and the features represent proximity, shape similarity, and closure between two components. In addition, changing combination of the features, plural grouping patterns are output. Besides, this paper shows the efficiency and limits of the proposed method from experimental results. In the experiments, 104 participants have perceived grouping patterns to 74 trademark images and the human perceptions have been compared with outputs by the proposed method for the 74 images.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.

Satellite Internet is one of the most important networks for emergency communications because of its tolerant of disasters such as earthquake. Therefore, satellite Internet has received considerable attention over recent years. However, most standard implementations of TCP congestion control method perform poorly in satellite Internet due to its high bit error rate and long propagation delay. This paper proposes a new TCP congestion control method called TCP-STAR to improve the throughput over satellite Internet. TCP-STAR has three new mechanisms, namely Congestion Window Setting (CWS) based on available bandwidth, Lift Window Control (LWC), and Acknowledgment Error Notification (AEN). CWS can resist the reduction of the transmission rate when data losses are caused by bit error. LWC is able to increase the congestion window quickly based on the estimated available bandwidth. AEN can avoid the reduction of the throughput by mis-retransmission of data. The mis-retransmission is caused by ack losses or delay. Simulations show that TCP-STAR can obtain the best throughput comparing with other TCP variants (TCP-J and TCP-WestwoodBR). Furthermore, we found that the fairness of TCP-STAR is a little lower than that of TCP-WestwoodBR. However, the fairness of TCP-STAR is equal to TCP-J.

For backward compatibility, ECN-capable networks should be capable of handling both ECN-capable and ECN-incapable TCP flows. In this letter, we present a backward congestion notification (BCN) scheme that can provide fast congestion indication delivery, while improving fairness between ECN-capable and ECN-incapable flows. Simulation results reveal that the BCN scheme is more effective than the original ECN mechanism in terms of stability, throughput, and fairness.

A topological book embedding of a graph is an embedding in a book that carries the vertices in the spine of the book and the edges in the pages so that edges are allowed to cross the spine. Recently, the author has shown that for an arbitrary graph G with n vertices there exists a d+1-page book embedding of G in which each edge crosses the spine logd n times. This paper improves the result for the case of bipartite graphs and shows that there exists a d+1-page book embedding of a bipartite graph Gn1,n2 having two partite sets with n1 and n2 vertices respectively (n1 ≥ n2) in which each edge crosses the spine logd n2 -1 times.

In this paper, a method is proposed to construct a visual secret sharing (VSS) scheme for multiple secret images in which each share can be rotated with 180 degrees in decryption. The proposed VSS scheme can encrypt more number of secret images compared with the normal VSS schemes. Furthermore, the proposed technique can be applied to the VSS scheme that allows to turn over some shares in decryption. From the theoretical point of view, it is interesting to note that such VSS schemes cannot be obtained from so-called basis matrices straightforwardly.

Network congestion and random errors of wireless link are two well-known noteworthy parameters which degrade the TCP performance over heterogeneous networks. We put forward a novel end-to-end TCP congestion control mechanism, namely TCP BaLDE (Bandwidth and Loss Differentiation Estimate), in which the TCP congestion control categorizes the reason of the packet loss by estimating loss differentiation in order to control the packet transmission rate appropriately. While controlling transmission rate depends on the available bandwidth estimation which is apprehended by the bandwidth estimation algorithm when the sender receives a new ACK with incipient congestion signal, duplicates ACKs or is triggered by retransmission timeout event. Especially, this helps the sender to avoid router queue overflow by opportunely entering the congestion avoidance phase. In simulation, we experimented under numerous different network conditions. The results show that TCP BaLDE can achieve robustness in aspect of stability, accuracy and rapidity of the estimate in comparison with TCP Westwood, and tolerate ACK compression. It can achieve better performance than TCP Reno and TCP Westwood. Moreover, it is fair on bottleneck sharing to multiple TCP flows of the same TCP version, and friendly to existing TCP version.

In this paper, based on Equivalent Active Flow, we propose a novel technique called Approximate Fairness Dropping, which is able to approximate fairness by containing misbehaving flows' access queue opportunity with low time/space complexity. Unlike most of the existing Active Queue Management schemes (e.g., RED, BLUE, CHOKE), Approximate Fairness Dropping does not drop the packets whose arriving rate is within the maximum admitted rate, so it protects the well-behaving flows against misbehaving ones, moreover, improves the throughput and decreases the queuing delay. Our simulations and analyses demonstrate that this new technique outperforms the existing schemes and closely approximates the "ideal" case, where full state information is needed.

Since congestion is very likely to happen in the Internet, locating congested areas (path segments) along a congested path is vital to appropriate actions by Internet Service Providers to mitigate or prevent network performance degradation. We propose a practical method to locate congested segments by actively measuring one-way end-to-end packet losses on appropriate paths from multiple origins to multiple destinations, using a network tomographic approach. Then we conduct a long-term experiment measuring packet losses on multiple paths over the Japanese commercial Internet. The experimental results indicate that the proposed method is able to precisely locate congested segments. Some findings on congestion over the Japan Internet are also given based on the experiment.

The most important design goal in Optical Burst Switching (OBS) networks is to reduce burst loss resulting from resource contention. Especially, the higher the congestion degree in the network is, the higher the burst loss rate becomes. The burst loss performance can be improved by employing a judicious congestion control. In this paper, to actively avoid contentions, we propose a peak load-based congestion control scheme that operates based on the highest (called peak load) of the loads of all links over the path between each pair of ingress and egress nodes in an OBS network. Simulation results show that the proposed scheme reduces the burst loss rate significantly, compared to existing OBS protocols, while maintaining reasonable throughput and fairness.

Lossless video coding is required in the fields of archiving and editing digital cinema or digital broadcasting contents. This paper combines a discrete wavelet transform and adaptive inter/intra-frame prediction in the wavelet transform domain to create multiresolution lossless video coding. Based on the image statistics of the wavelet transform domains in successive frames, inter/intra frame adaptive prediction is applied to the appropriate wavelet transform domain. This adaptation offers superior compression performance. A progressive transmission scheme is also proposed for effective resolution scalability. Experiments on test sequences confirm the effectiveness of the proposed algorithm.

Popular Web sites form their Web servers into Web server clusters. The Web server cluster operates with a load-balancing algorithm to distribute Web requests evenly among Web servers. The load-balancing algorithms founded on conventional periodic load-information update mechanism are not scalable due to the synchronized update of load-information. We propose a load-balancing algorithm that the load-information update is not synchronized by exploiting variant execution times of executing scripts in dynamic Web pages. The load-information of each server is updated 'individually' by a new load-information update mechanism, and the proposed algorithm supports high scalability based on this individual update. Simulation results have proven the improvement in system performance through another aspect of high scalability. Furthermore, the proposed algorithm guarantees some level of QoS for Web clients by fairly distributing requests. A fundamental merit of the proposed algorithm is its simplicity, which supports higher throughput of the Web switch.

In this letter we examine two transport protocols, HighSpeed TCP [1] and Scalable TCP [2] which are both sender-side varieties of TCP. Based on the fluid flow theory, we develop a general nonlinear model and use gain margin and phase margin to evaluate the stability of a closed-loop system which is composed of a transport protocol and an active queue management scheme. Our results indicate that HSTCP and STCP are stabler than standard TCP when link bandwidth, flow number and round-trip time vary.

This paper introduces a network-aware video delivery framework where the quality-of-service (QoS) interaction between prioritized packet video and relative differentiated service (DiffServ) network is taken into account. With this framework, we propose a dynamic class mapping (DCM) scheme to allow video applications to cope with service degradation and class-based resource constraint in a time-varying network environment. In the proposed scheme, an explicit congestion notification (ECN)-based feedback mechanism is utilized to notify the status of network classes and the received service quality assessment to the end-host applications urgently. Based on the feedback information, DCM agent at ingress point can dynamically re-map each packet onto a network class in order to satisfy the desired QoS requirement. Simulation results verify the enhanced QoS performance of the streaming video application by comparing the static class-mapping and the class re-mapping based on loss-driven feedback.

The queue size at a bottleneck would impact the performance of TCP protocols, especially when running a single TCP flow in networks with a large bandwidth-delay product. However, queue size has been not well considered in experiments. This paper shows how bottleneck queue size influences TCP protocols performance. Bursityness of advanced TCPs is examined. Ways of estimating queue size are introduced. Sending a UDP packet train until a loss is detected is a method to measure queuing delay to estimate queue size. Watching a loss in a TCP session to measure round trip time and calculate the queue size is also discussed. Results from experiments with a network emulator and a real network are reported. The results indicated that a layer-2 switch at a congestion point would be a major factor of decreasing TCP performance in a fast long distant path.

Mobile devices have acceleratedly penetrated into our daily lives. Though they were originally designed as a communication tool or for personal use, and due to the rapid availability of wireless network technologies, people have begun to use mobile devices for supporting collaborative work and learning. There is, however, a serious problem in mobile devices related to their user interfaces. In this paper, we try to alleviate the problem and propose intuitive techniques for information transfer, which is one of the typical usages of mutually-connected computers. Our system, Toss-It, enables a user to send information from the user's PDA to other electronic devices with a "toss" or "swing" action, like a user would toss a ball or deal cards to others. The implementation of Toss-It consists of three principle parts - gesture recognition, location recognition, and file transfer. We especially describe the details of gesture recognition and location recognition. We then evaluate the practicability and usability of Toss-It through the experiments. We also discuss user scenarios describing how Toss-It can support users' collaborative activities.

During devastating natural disasters, numerous people want to make calls to check on their families and friends in the stricken areas, but many call attempts on mobile cellular systems are blocked due to limited radio frequency resources. To reduce call blocking and enable as many people as possible to access mobile cellular systems, placing a limit on the holding time for each call has been studied [1],[2]. However, during a catastrophe, emergency calls, e.g., calls to fire, ambulance, or police services are also highly likely to increase and it is important that the holding time for these calls is not limited. A method of limiting call holding time to make provision for emergency calls while considering the needs of ordinary callers is proposed. In this method, called the HTL-E method, all calls are classified as emergency calls or other according to the numbers that are dialed or the terminal numbers that are given in advance to the particular terminals making emergency calls, and only the holding time of other calls is limited. The performance characteristics of the HTL-E method were evaluated using computer simulations. The results showed that it reduced the rates of blocking and forced call termination at handover considerably, without reducing the holding time for emergency calls. The blocking rate was almost equal for emergency and other calls. In addition, the HTL-E method handles fluctuations in the demand for emergency calls flexibly. A simple method of estimating the holding-time limit for other calls, which reduces the blocking rate for emergency and other calls to the normal rate for periods of increased call demand is also presented. The calculated results produced by this method agreed well with the simulation results.

It has been a very important issue to evaluate the performance of transmission control protocol (TCP), and the importance is still growing up because TCP will be deployed more widely in future wireless as well as wireline networks. It is also the reason why there have been a lot of efforts to analyze TCP performance more accurately. Most of these works are focusing on overall TCP end-to-end throughput that is defined as the number of bytes transmitted for a given time period. Even though each TCP's fast recovery strategy should be considered in computation of the exact time period, it has not been considered sufficiently in the existing models. That is, for more detailed performance analysis of a TCP implementation, the fast recovery latency during which lost packets are retransmitted should be considered with its relevant strategy. In this paper, we extend the existing models in order to capture TCP's loss recovery behaviors in detail. On the basis of the model, the loss recovery latency of three TCP implementations can be derived with considering the number of retransmitted packets. In particular, the proposed model differentiates the loss recovery performance of TCP using selective acknowledgement (SACK) option from TCP NewReno. We also verify that the proposed model reflects the precise latency of each TCP's loss recovery by simulations.

This paper proposes an efficient technique for human motion recognition based on motion history images and an eigenspace technique. In recent years, human motion recognition has become one of the most popular research fields. It is expected to be applied in a security system, man-machine communication, and so on. In the proposed technique, we use two feature images and the eigenspace technique to realize high-speed recognition. An experiment was performed on recognizing six human motions and the results showed satisfactory performance of the technique.