Arc duration of silver contacts was measured with fixed short gap under the condition of different constant gaps and different constant currents, which are close to the minimum arc current of the contact material. The results indicated that the arc occurred even at the current less than the minimum arc current. The arc duration varies in different operations at the same testing condition and it distributes exponentially in the time range up to 100 microseconds. The time constant τ and the extinction rate λ of the distribution were calculated, and related to the arc current. It was assured by silver electrode arc that, even below the so-called minimum arc current, arc ignition may occur but, in this case, the arc duration is momentary, and, as the arc current decreases, the arc extinction rate becomes large.

To support UDP-based real-time multimedia applications over the Internet, it is necessary to provide a certain amount of bandwidth within the network so that the performance of the applications will not be seriously affected during periods of congestion. Since the flow rates of some of these applications do not back off during periods of congestion, it is also necessary to protect flow-controlled TCP flows from unresponsive or aggressive UDP flows. To achieve these goals, we propose a simple queue policy to support multimedia applications, called threshold-based queue management (TBQM). TBQM isolates UDP flows efficiently from TCP flows to protect TCP flows while supporting bandwidth requirements of UDP applications that require QoS. In addition, TBQM supports drop fairness between flows without maintaining per-flow state. We also present some experimental results to show that the proposed queue policy works well.

Takine investigated the continuous time queueing system with single server and Markovian arrival streams (MAS) where service times of customers depend on the states of the underlying Markov chain immediately before and after arrivals and may differ from one another among streams. In this paper, we consider the discrete version of Takine's model. Since a MAS can be used to model the correlated traffics of several classes, we consider the discrete time queueing system with MAS as an input traffic. Moreover packets from different classes of traffic may require different service times. So service times of customers which are different from one another among streams and may depend on the states of the underlying Markov chain immediately before and after arrivals, are also considered. We obtain the distribution of the workload in the system using Neuts' matrix analytic method. Based on this result, we obtain the distributions of the actual waiting time and the sojourn time of a customer of each class and the joint distribution of the numbers of customers of each class. We also provide a recursion formula for computing the joint queue length distribution. We give an application to multimedia conference system and numerical example.

In this paper, we present an Internet personal live-broadcasting server system. Our solution is not only for experts but also for amateur users who want to broadcast using simple multimedia equipment. For scalable broadcasting services, we developed multiple-channel establishment and channel expansion. Concurrent services for a large number of broadcasting channels are effectively provided. Also, the capacity of channels can be expanded as the number of participants increases. Furthermore, for the sake of complete live broadcasting with high-quality transmission, the system supports both TCP (transmission control protocol) and UDP (user datagram protocol) according to the status of network environments as well as the received packet loss in the user system. The performance of the system is effectively evaluated at such practical commercial sites as well-known community and E-business sites.

In this paper we present reconfigurable computing as a compelling choice of computing platform for real-time, onboard processing for satellite applications. In particular, we discuss the use of reconfigurable computing in the context of a real-time remote sensing system, providing motivation for such a system and describing attributes of reconfigurable computing that support it as the technology of choice. The High Performance Computing (HPC-I) payload, designed and developed for the Australian scientific satellite FedSat, is introduced as a demonstration of onboard processing in space using reconfigurable logic. We present an overview of the real-time remote sensing system architecture, and describe the design and implementation of three remote sensing algorithms in HPC-I for cloud masking, wildfire detection and volcanic plume detection. Finally, results from simulation and testing are presented which show very promising performance in terms of data throughput and detection capabilities.

We designed an FPGA-based parallel machine called "RASH"(Reconfigurable Architecture based on Scalable Hardware) for high speed and flexible signal/data processing. Cryptanalysis is one of the killer applications for FPGA-based machines because huge amounts of logical and/or simple arithmetic operations are required and FPGA is suitable for this. One of the well-known activities in cryptanalysis is the DES (Data Encryption Standard) cracking contest conducted by RSA Data Security. TMTO (Time-Memory Trade-Off) Cryptanalysis is a practical method to dramatically shorten the time for key search when plaintext is given in advance. A string of ASCII characters is used as the key much like a password. The ASCII character is 7-bit character and is changed to 96 kinds of value. The 56-bit DES key is given with a string of 8 ASCII characters. Although the DES key has 64 trillion(=256) possibilities, the key that is given with a string has only 6.4 trillion(=968) possibilities. Therefore, we improve TMTO cryptanalysis so that we search only the limited key by ASCII characters and reduce the quantity of computation. In this paper, we demonstrate how TMTO cryptanalysis for limited key is well suited to our FPGA-based RASH machine. By limiting the key to a string, DES key will be found at 80% probability within 45 minutes after ciphertext is given on 10 units of RASH. The precomputation before starting key search takes 3 weeks on the same RASH configuration.

We propose a new algorithm for blind source separation (BSS), in which frequency-domain independent component analysis (FDICA) and time-domain ICA (TDICA) are combined to achieve a superior source-separation performance under reverberant conditions. Generally speaking, conventional TDICA fails to separate source signals under heavily reverberant conditions because of the low convergence in the iterative learning of the inverse of the mixing system. On the other hand, the separation performance of conventional FDICA also degrades significantly because the independence assumption of narrow-band signals collapses when the number of subbands increases. In the proposed method, the separated signals of FDICA are regarded as the input signals for TDICA, and we can remove the residual crosstalk components of FDICA by using TDICA. The experimental results obtained under the reverberant condition reveal that the separation performance of the proposed method is superior to those of TDICA- and FDICA-based BSS methods.

In this letter, we propose a frequency domain active noise control system using the time difference simultaneous perturbation method. This method is an algorithm based on the simultaneous perturbation method which updates the coefficients of the noise control filter only by use of the error signal. The time difference simultaneous perturbation method updates the filter coefficients by using one kind of error signal, while the simultaneous perturbation method updates the filter coefficients by using two kinds of error signal. In the ANC systems, the time difference simultaneous perturbation method is superior because ANC systems cannot obtain two error signals at the same time. When this method is applied to ANC systems, the convergence speed can be increased to a maximum of twice that of the conventional method.

The 2-vertex-connectivity augmentation problem for specified vertices (2VCA-SV) is defined as follows: Given an undirected graph G=(V,E), a subgraph G0=(V,E') of G, a specified set of vertices S V and a weight function w:E R^+ (nonnegative real numbers), find a set E" E-E' with the minimum total weight, such that G0+E"=(V,E' E") has at least two internally disjoint paths between any pair of vertices in S. In this paper, we propose an O(|V||E|+ |V|2 log |V|) time algorithm 2-ABIS, whose performance ratio is 2 (3, respectively), for 2VCA-SV if G0 has a connected component containing S (otherwise).

A 12.8GOPS/2.1GFLOPS that operates at 533 MHz, and which is equipped with an 8-way embedded VLIW processor fabricated with 6-layer Cu and 1-layer Al metal 0.11 µm CMOS process technology is introduced in this paper. The processor is also equipped with a 4-way integer pipeline, a 4-way floating/media pipeline, and separate 32 KB 4-way set associative instruction and data caches. It is also equipped with instruction fetch prediction, and non-aligned dual data load/store mechanisms. The performance evaluation that was successfully conducted using the MPEG2 IDCT routine and JPEG decoding program shows that it offers twice the performance of the previous work . 10.4 M transistors are integrated on a 7.8 mm 7.8 mm die. The chip consumes 2.5 W at 533 MHz.

The proposed medium access protocol deals especially with the timely-transmission of real-time packets in wireless multimedia networks where users of many types of traffic are present. It works based on Time Division Multiple Access/Time Division Duplex (TDMA/TDD) technique and fixed-length packet switching incorporating two different policies to work differently on either non-congestion or congestion periods. In the policy to deal with congestion periods the concept of urgent packet has been introduced as any packet whose transmission deadline is on the next frame. Hence, during periods of congestion users inform to the Base Station the number and average deadline of the urgent packets in their buffers through requirement messages. According to that information the system is able to distribute its resources in a more efficient way during periods of congestion making the real-time packet loss rate decrease considerably. The simulation results show a very good performance of the method in networks where different types of traffic coexist even under high traffic-load conditions. The results also show a good trade-off characteristic between the real-time access delay and the buffer occupancy of non-real time terminals during congestion periods.

Combining a layered space-time receiver with the OFDM for high-rate transmissions requires the multi-channel estimation process. Therefore, this letter highlights a design of a preamble structure for OFDM-based layered space-time receiver. The new proposed preamble can estimate the multi-channel up to the 8 transmit antennas using two long preambles in the WLAN standards.

The design criteria for space-time trellis codes (STTC's) in fast fading channels have been proposed: the Distance Criterion and the Product Criterion. The design criteria in [1] are based on optimizing the pairwise error probability (PWEP). However, the frame error rate (FER) of STTC's depends on the distance spectrum. In this paper, we propose a new design criterion for STTC's based on the distance spectrum in fast fading channels. The proposed design criterion is based on the product distance distribution for the large signal-to-noise ratio (SNR) and the trace distribution for the small SNR, respectively. Moreover, we propose new STTC's by the computer search based on the proposed design criterion in fast fading channels. By computer simulation, we show that the proposed design criterion is more useful than the Product Criterion in [1] in fast fading channels. We also show that the proposed STTC's achieve better FER than the conventional STTC's in fast fading channels.

On research to determine reliable acoustic indicators for the type of stress present in speech, the majority of systems have concentrated on the statistics extracted from pitch contour, energy contour, wavelet based subband features and Teager-Energy-Operator (TEO) based feature parameters. These systems work mostly on pair-wise distinction between stress and neutral speech. Their performance decreases substantially when tested in multi-style detection among many stress categories. In this paper, a novel system is proposed using linear short time Log Frequency Power Coefficients (LFPC) and TEO based nonlinear LFPC features in both time and frequency domain. Five-state Hidden Markov Model (HMM) with continuous Gaussian mixture distribution is used. The stress classification ability of the system is tested using data from the SUSAS (Speech Under Simulated and Actual Stress) database to categorize five stress conditions individually. It is found that the performance of linear acoustic features LFPC is better than that of nonlinear TEO based LFPC feature parameters. Results show that with linear acoustic feature LFPC, average accuracy of 84% and the best accuracy of 95% can be achieved in the classification of the five categories. Results of test of the system under different signal-to-noise conditions show that the performance of the system does not degrade drastically with increase in noise. It is also observed that classification using nonlinear frequency domain LFPC features gives relatively higher accuracy than that using nonlinear time domain LFPC features.

The problem of optimal placement of pilot symbols is considered for single carrier packet-based transmission over time varying channels. Both flat and frequency-selective fading channels are considered, and the time variation of the channel is modeled by Gauss-Markov process. The semi-blind linear minimum mean-square error (LMMSE) channel estimation is used. Two different performance criteria, namely the maximum mean square error (MSE) of the channel tap state over a packet and the cumulative channel MSE over a packet, are used to compare different placement schemes. The pilot symbols are assumed to be placed in clusters of length (2L+1) where L is the channel order, and only one non-zero training symbols is placed at the center of each cluster. It is shown that, at high SNR, either performance metric is minimized by distributing the pilot clusters throughout the packet periodically. It is shown that at low SNR, the placement is in fact not optimal. Finally, the performance under the periodic placement is compared with that obtained with superimposed pilots.

A two-qubit system made of electrons running along coupled pairs of quantum wires is described and numerically analyzed. A brief review of the basic gates is given first, based on preliminary investigations, followed by the description of the electron dynamics. A detailed analysis of a conditional phase shifter is carried out by means of a time-dependent Schrodinger solver applied to a two-particle system. A quantum network suitable for creating entanglement is simulated, and results are shown. The physical structure of the proposed network is within the reach of a solid-state implementation. The physical parameters used in the computations have been chosen with reference to silicon quantum wires embedded in silicon dioxide.

This paper defines the distributed arrangement selection problem in a line network in a distributed context and describes the design of a strictly-time-optimal algorithm which solves the problem with a limited local memory space. The problem is regarded as a combined distributed sorting and k-selection problem, namely a problem of sorting elements that are not larger than the kth minimum element in predetermined processes. The algorithm also provides a solution to a resource allocation problem in a line network in a strictly-optimal time.

Unicasting video streams over TCP connections is a challenging problem, because video sources cannot normally adapt to delay and throughput variations of TCP connections. This paper describes a method of extending TCP so that TCP connections can effectively carry hierarchically-encoded layered video streams, while being friendly to other competing connections. We call the method Receiver-based Delay Control (RDC). Under RDC, a TCP connection can slow down its transmission rate to avoid congestion by delaying ACK packet generation at the TCP receiver based on congestion notifications from routers. We present the principle behind RDC, argue that it is TCP-friendly, describe an implementation that uses 1-bit congestion notification from routers, and demonstrate by simulations its effectiveness in streaming hierarchically-encoded layered video.

In the past, enhancement techniques for the end-to-end quality of a networked application were studied by looking at each individual layer. Examples of such techniques include the error resilience/concealment methods in the application layer, the FEC/ARQ in the transport layer, and the Quality of Service (QoS) techniques in the network layers. However, an integrated approach that would look across all related layers had yet to be investigated. This paper proposes an approach that combines priority-aware video packetization, adaptively used layered FEC, and QoS controlled networks such as IntServ and DiffServ in order to provide an efficient end-to-end quality in layered streaming video. The combination is more efficient in terms of a simple network price mechanism, that is, in getting the best end-to-end quality under a given total cost constraint. Our proposed approach in DiffServ with video packet (VP) data-splitting and layered FEC guarantees minimal service quality in a scalable and cost effective manner without introducing resource reservation. For video, we also propose performance metrics such as corrupted/frozen frame ratio (CFR, FFR). These provide objective metrics like PSNR as well as a measurement for subjective perceptions. Our approach, which combines related layers such as video coding, transport, and network, has yielded results that have proven to be more cost-effective and practical than the supporting network QoS.

This paper introduces a way to realize high-pass, band-stop and all-pass transfer functions using two-integrator loop structure consisting of loss-less and lossy integrators. The basic circuit configuration is constructed with five Operational Transconductance Amplifiers (OTAs) and two grounded capacitors. It is shown that the circuit can realize their circuit transfer functions by choosing the input terminals, and that the circuit parameters can also be independently set by the transconductance gains with the proportional block. Although the basic circuit configuration has been known, it seems that the feature for realizing the high-pass, the band-stop and the all-pass transfer functions makes the structure more attractive and useful. An example is given together with simulated results by PSPICE.