This paper describes a new optical label switching technique; wavelength and pilot tone frequency are combined to form labels that are used to control transport network routing. This technique is very attractive for achieving simple nodes that offer extremely rapid forwarding. Experimental results on the discrimination of optical labels and all-optical label conversion are also presented.

We consider methods for threshold RSA decryption among distributed agencies without any dealer or trusted party. The first solution is a combination of two techniques by [9] and [7] . It demonstrates the feasibility of combining the distributed key generation and the RSA secure function application. The second solution is another approach making the distributed key distribution simpler and alleviating a burden of each shareholder in comparison with the first scheme. The latter scheme is newly developed technique based on [9] and further inspired by Simmons' protocol-failure of RSA (we believe that it is very interesting that a "protocol failure attack" be turned into a constructive method). Our comparison between these two schemes indicates a new measure of the performance of a distributed cryptographic protocol that consists of multiple stages.

This paper addresses the problem of detecting digital line components in a given binary image consisting of n black dots arranged over N N integer grids. The most popular method in computer vision for this purpose is the one called Hough Transform which transforms each black point to a sinusoidal curve to detect digital line components by voting on the dual plane. We start with a definition of a line component to be detected and present several different algorithms based on the definition. The one extreme is the conventional algorithm based on voting on the subdivided dual plane while the other is the one based on topological walk on an arrangement of sinusoidal curves defined by the Hough transform. Some intermediate algorithm based on half-planar range counting is also presented. Finally, we discuss how to incorporate several practical conditions associated with minimum density and restricted maximality.

Among several multiprocessor topologies, two-dimensional (2D) mesh topology has become popular due to its simplicity and efficiency. Even though a number of scheduling and processor allocation schemes for 2D meshes have been proposed in the literature, little study has been done aimed for real-time environment. In this paper, we propose an on-line scheduling and allocation scheme for real-time tasks that require the exclusive use of submeshes in 2D mesh system. By effectively manipulating the information on allocated or reserved submeshes, the proposed scheme can quickly identify the earliest available time of a free submesh for a newly arrived task. We employ a limited preemption approach to reduce the complexity of the search for a feasible schedule. Computer simulation reveals that the proposed scheme allows high throughput by decreasing the number of tasks rejected.

A K-band MMIC subharmonically pumped mixer integrating local oscillator (LO) amplifier has been developed. For up-converter application, it is necessary to reduce the leakage of second harmonic component of LO frequency to RF port, which is generated by nonlinear operation of LO amplifier. A quasi-lumped short-circuited stub using microstrip structure has been successfully applied to the MMIC mixer to enhance 2fLO-suppression. We propose a new configuration of a quasi-lumped short-circuited stub, which reduces the influence of parasitic elements of via-holes. The developed MMIC has a one-stage LO amplifier and it has shown about 10 dB-improvement of 2fLO-suppression compared to conventional configuration using a quarter-wavelength short-circuited stub.

Mass measurement system for the measurement of mass of substances placed in randomly vibrating circumstances has been developed. Mass measurement range was defined from 0 g to 400 g for the primary model with the measurement error of approximately 3% when randomly directional vibration of 6 m/sec2 acceleration was applied to the system.

One of the disadvantages of compressed data is their vulnerability, that is, even a single corrupted bit in compressed data may destroy the decompressed data completely. Therefore, Variable-to-Fixed length Arithmetic Coding, or VFAC, with error detecting capability is discussed. However, implementable error recovery method for compressed data has never been proposed. This paper proposes Burst Error Recovery Variable-to-Fixed length Arithmetic Coding, or BERVFAC, as well as Error Detecting Variable-to-Fixed length Arithmetic Coding, or EDVFAC. Both VFAC schemes achieve VF coding by inserting the internal states of the decompressor into compressed data. The internal states consist of width and offset of the sub-interval corresponding to the decompressed symbol and are also used for error detection. Convolutional operations are applied to encoding and decoding in order to propagate errors and improve error control capability. The proposed EDVFAC and BERVFAC are evaluated by theoretical analysis and computer simulations. The simulation results show that more than 99.99% of errors can be detected by EDVFAC. For BERVFAC, over 99.95% of l-burst errors can be corrected for l 32 and greater than 99.99% of other errors can be detected. The simulation results also show that the time-overhead necessary to decode the BERVFAC is about 12% when 10% of the received words are erroneous.

A new speech enhancement technique is proposed assuming that a speech signal is represented in terms of a linear probabilistic process and that a noise signal is represented in terms of a stationary random process. Since the target signal, i.e., speech, cannot be represented by a stationary random process, a Wiener filter does not yield an optimum solution to this problem regarding the minimum mean variance. Instead, a Kalman filter may provide a suitable solution in this case. In the Kalman filter, a signal is represented as a sequence of varying state vectors, and the transition is dominated by transition matrices. Our proposal is to construct the state vectors as well as the transition matrices based on time-frequency pattern of signals calculated by a wavelet transformation (WT). Computer simulations verify that the proposed technique has a high potential to suppress noise signals.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

In this paper, we first investigate the problems of the existing branch point algorithms for available bit rate (ABR) multicast connections in ATM networks, and then propose various solutions for resolving those problems. By combining these solutions, we also propose a new efficient and scalable branch point algorithm. In the proposed algorithm, each branch point stores the feedback information on a per-branch basis for each virtual connection and only passes BRM cells returning from the farthest destination. Simulation results show that the proposed algorithm can provide a good fairness, a higher efficiency and an excellent scalability, compared with the existing algorithms.

This paper describes the evaluation of a fiber-optic reflective displacement sensor that is compensated for variations in light source intensity, pressure, temperature and opacity of ambient medium. Additionally, the distance information is averaged over several points on the target surface, which reduces signal fluctuations due to inhomogeneities. Furthermore, a practical optical fiber reflective sensor model of measuring oil film thickness for thrust bearing is set up in this paper. Actual measurements were made with HEC 3000 tons' thrust bearing and the results were in good agreement with theoretical calculations.

In this paper, we propose a new location tracking method for pointer forwarding with distributed HLRs schemes to decrease the location tracking cost. Similar to the distributed HLR schemes, the location information is replicated on all HLRs in our scheme. However, unlike the traditional distributed HLR schemes, we propagate the most recent location changes from an HLR to reset the forwarding pointers in other HLRs, and thus reduce the cost of location tracking for call deliveries. The signaling cost of location propagation may be too costly to be practical. Therefore, we allow some inconsistency to exist among the location data of HLRs and defer the propagation of location changes to reduce the propagation cost. Three delay propagation policies are also proposed and analyzed using simulation. The performance result shows that the delay propagation policies reduce the propagation cost and the propagation resetting process degrades the location tracking cost. Our method preserves a low total cost for call deliveries, registrations and propagations in distributed HLRs.

We examine delay performance of packets from constant bit rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the Σ Di/G/1 queue with vacations. Our analysis allows heterogeneous service time and heterogeneous interarrival time. Thus, we can get the impact of packet length of a stream on the delay time of other streams. We then give various numerical results for enterprise multimedia networks, which include voice, video and data communication services. From our quantitative evaluation, we conclude that packet length of video traffic has large influence on the delay time of voice traffic while voice traffic gives a little impact on the delay time of video traffic.

Differentiated services (Diffserv) model is one of the possible solutions for providing quality of service (QoS) on the Internet. Most existing approaches assume that the packet loss is an indication of network congestion and thus reduce the sending rates of sources. For wireless networks, the assumption is not proper since packet losses may be caused by other reasons, such as fading and interference of the signal. Therefore, these approaches do not work well in wireless networks. In this paper, we propose an approach which is able to provide service differentiation in wireless environments. In our approach, the rate share of a connection is determined by the associated weight. By keeping a proper amount of extra data in the network, the proposed approach can achieve weighted proportional fairness, which can provide selective QoS without any particular support from the network. We use the ns simulator to evaluate our approach. Simulation results show the validity of the proposed approach.

Due to the lack of support in mobility and quality of service, today's IP-based networks have some inherent limitations for delivering multimedia services in a mobile environment. In the past few years, these issues have been addressed in the research community and the resulting techniques are being standardised. However, these developments have been done in isolation and become incompatible with each other. In addition to these technical issues, the future infrastructure for charging and accounting mobile multimedia services is expected to be increasingly complicated. In this paper we present a home-proxy based framework which can facilitate the integration of mobility support and QoS management. Furthermore, it enables centralised accounting, which simplifies the cost recovery processes of roaming services. To prove the viability of our design, we have built and tested this framework on a DiffServ wide-area backbone using an MP3 streaming application.

Error concealment is an essential part of reliable video communication systems because transmission errors are inevitable even when the coded bitstream is highly protected. The problem of temporal EC can be factored into two parts regarding candidate motion vectors (MVs) employed and the matching criterion to evaluate the fitness of each candidate MV. In order to obtain more faithful EC results, this paper proposes a novel iterative EC algorithm, in which an efficient way to provide candidate MVs and a new fitness measure are presented. The proposed approach for candidate MVs systematically utilizes all the available neighboring MVs by exploiting a well-known spatiotemporal correlation of block MVs. Also, in order to remove the dependency of a damaged block's quality of concealment on the already concealed adjacent blocks, we develope a new matching criterion. The objective of the proposed fitness measure is to minimize the total boundary matching errors induced by the whole corrupted blocks. Simulations performed using an H.263 codec demonstrate a significant improvement on the subjective and objective concealed video qualities, especially when the corrupted area is wider than a single row of coding blocks.

In this paper, a medium access control protocol is proposed for the integrated voice and data services in the local CDMA communication systems. Based on WB-TD/CDMA, uplink channels for the proposed protocol are composed of time slots with multiple spreading codes for each slot. During a talkspurt, a voice terminal transmits its entire packets over a reserved code. On the other hand, a data terminal transmits its packet after sensing the spreading code status. The base station broadcasts the status of spreading codes for each slot. In this protocol, voice packets never contend with data packets. The numerical results show that this protocol increases the system capacity for voice service by applying the reservation scheme. Despite the low access priority of data terminal, the data traffic performance can be increased in proportion to the number of spreading codes.

Since the detection of optical flow (two-dimensional motion field on an image) from image sequences is essentially an ill-posed problem, most of the conventional methods use a smoothness constraint for optical flow heuristically and detect reasonable optical flow. However, little discussion exists regarding the degree of smoothness. Furthermore, to recover the relative three-dimensional motion and depth between a camera and a rigid object, in general at first, the optical flow is detected without a rigid motion constraint, and next, the motion and depth are estimated using the detected optical flow. Rigorously speaking, the optical flow should be detected with such a constraint, and consequently three-dimensional motion and depth should be determined. To solve these problems, in this paper, we apply a parametric model to an optical flow, and construct an estimation algorithm based on this model.

This paper describes a new method for estimating the direction of arrival (DOA) using a nonlinear microphone array system based on complementary beamforming. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, since the resultant directivity pattern is proportional to the product of these directivity patterns, the proposed method can be used to estimate DOAs of 2(K-1) sound sources with K-element microphone array. First, DOA-estimation experiments are performed using both computer simulation and actual devices in real acoustic environments. The results clarify that DOA estimation for two sound sources can be accomplished by the proposed method with two microphones. Also, by comparing the resolutions of DOA estimation by the proposed method and by the conventional minimum variance method, we can show that the performance of the proposed method is superior to that of the minimum variance method under all reverberant conditions.

The third generation cellular system will provide high bandwidth data service and multimedia applications are proposed on the high-speed data link. On the other hand, cellular handsets are getting equipped with a short-range radio communication device that is mainly targeted to connect cellular handsets to computers. In this paper, we propose a percolating data delivery mechanism for a cellular-ad hoc integrated network utilizing multicast/broadcast communication, which is endorsed by data synchronization, and single-hop ad hoc networks for information shower/exchange services. The mechanism lessens data traffic in both the cellular and the ad hoc network for data delivery and tolerates unexpected disconnection in the ad hoc network. The mechanism also compensates data delivery in areas out of the cellular service. Our implementation of the data delivery mechanism utilizes Bluetooth for an ad hoc network configuration, and delivers data in the cellular network to the Bluetooth ad hoc network via the data synchronization mechanism. We evaluate communication traffic and delivery time on the prototype system and discuss about the efficiency of the mechanism.