We study some sufficient conditions of codeword lengths for the existence of a fix-free code. Ahlswede et al. proposed the 3/4 conjecture that Σi=1n a-li 3/4 implies the existence of a fix-free code with lengths li when a=2 i. e. the alphabet is binary. We propose a more general conjecture, and prove that the upper bound of our conjecture is not greater than 3/4 for any finite alphabet. Moreover, we show that for any a2 our conjecture is true if codeword lengths l1,l2,. . . consist of only two kinds of lengths.

We propose a traffic measurement system which uses trap and polling methods. To obtain its performance we consider a queueing model with a single server and evaluate a packet delay. In our multi-cast traffic, packets are modeled as a batch with a batch size distribution {gk}. The batch arrival process is observed as two processes on the basis of batch size. For a batch whose size is more than or equal to a threshold L, the batch will be trapped by our traffic measurement system (in queueing model, it will enter a queue immediately after its arrival). For a batch whose size is less than L, it will be observed at a polling cycle T (in queueing model, it will be temporarily stored in a buffer and all these small batches will be cyclically noticed with a cycle T). We analyze this queueing model by a diffusion approximation and compare the packet delay observed by our traffic measurement system with the L=1 original batch arrival model. Evaluating the results of the diffusion approximation, we illustrate that our traffic measurement system has functions not only to give an accurate estimation of the mean waiting time but also reduce the number of measurements by choosing appropriate parameters L and T.

This paper reports on time-space conversion-based differential processing of optical signals using a high-resolution arrayed-waveguide grating (AWG) and a spatial filter at a wavelength of 1.55 µm. We clarify the advantages of the AWG device and show where it is applicable. In order to reduce loss at the spatial filter, we propose a new phase-only filter that functions as a differential filter. The difference between the exact differential filter and the proposed phase-only filter is calculated theoretically. We confirm experimentally that the optical pulse can be differentiated by the proposed filter. For application of differential processing, we also proposed a phase modulation to amplitude modulation (PM-AM) conversion and demonstrated the PM-AM conversion at 10 Gbit/s signals using a PSK-non-return-to-zero (NRZ) format.

This paper reports on time-space conversion-based differential processing of optical signals using a high-resolution arrayed-waveguide grating (AWG) and a spatial filter at a wavelength of 1.55 µm. We clarify the advantages of the AWG device and show where it is applicable. In order to reduce loss at the spatial filter, we propose a new phase-only filter that functions as a differential filter. The difference between the exact differential filter and the proposed phase-only filter is calculated theoretically. We confirm experimentally that the optical pulse can be differentiated by the proposed filter. For application of differential processing, we also proposed a phase modulation to amplitude modulation (PM-AM) conversion and demonstrated the PM-AM conversion at 10 Gbit/s signals using a PSK-non-return-to-zero (NRZ) format.

An access attempt overload control method is proposed for CDMA cellular system to provide good communication quality to mobile stations under the traffic overload condition. Under the normal traffic condition, the effect of access attempts on the interference is negligible. But under the traffic overload condition, the interference due to access attempts could be large and deteriorate the communication quality (or reduce the capacity of a reverse link) if access attempts are not controlled properly. The numerical and simulation results show that our access attempt overload control method can maintain the interference caused by access attempts very low and the capacity of the reverse link is not reduced under the traffic overload condition.

A novel millimeter-wave on-wafer CAT(Computer-Aided-Testing ) system has been developed for measurement of S-parameters and NF ( Noise figure ). For the S-parameter test system, we have developed a holder setup and installed it in a semi-automatic wafer prober so that the waveguide-based T/R module can be directly connected to a probe-head through fixed waveguides, which feature low insertion loss of less than 2 dB, from 75 GHz to 98 GHz. The accuracy of the developed test system was confirmed by measuring, with this system, a co-planar offset short pattern then comparing measured and simulated results. A good agreement between the measured and calculated, in both return loss and return phase successfully demonstrated the superiority of the system. A W-band NF test system with a system noise of less than 8 dB has been also developed to provide an on-wafer NF measurement capability with an accuracy of 0.3 dB. These S-parameter and NF test systems possess great advantages to achieve high-speed automatic MMIC testing up to W-band.

Our simulation method by using a combination of discrete event simulation and detailed parametric models of the VLSI manufacturing test system is verified by comparing simulated results with actual ones of a real wafer test facility of one-chip microcomputer in a Japanese semiconductor company. The simulated results are found to be in close agreement with the actual ones. As an application of the verified simulation method, we evaluate the economic effect of the introduction in the wafer test process of LSI testers that allows us to test multiple chips simultaneously. It is found that the optimum number of chips simultaneously tested by an LSI tester is 4 when considering both of the test cost per chip and the average test TAT.

Future cellular systems are envisioned to support mixed traffic, and ultimately multimedia services. However, a mixture of voice and data requires novel service mechanisms that can guarantee quality of service. In order to transfer high-speed data, multislot channel allocation is seen as a favoured solution to the present systems with the least compromise to circuit- switched services. This paper evaluates the performance of narrowband voice calls and multislot data packet transmission in such integrated systems by using a matrix-analytic approach. This method achieves quadratic convergence compared to the conventional spectral methods. Mobility is also considered in a prioritized cellular environment where frequent handoff has the potential of degrading data performance. The voice call distribution, data packets throughput, delay and waiting time distribution are derived. Moreover, a new multiple priority-based distributed control algorithm and a voice rate control scheme are enforced to mitigate the queuing congestion of data packets. The numerical results derived from this study show that larger data packets incur longer latency and the use of these flexible schemes can improve the overall performance.

From our previous studies, we derived the worst case cell delay within an ATM switch and thus can find the worst case end-to-end delay for a set of real-time connections. We observed that these delays are sensitive to the priority assignment of the connections. With a better priority assignment scheme within the switch, the worst case delay can be reduced and provide a better network performance. We extend our previous work on the closed form analysis to conduct more experimental study of how different priority assignments and system parameters may affect the performance. Furthermore, from our worst case delay analysis on a regulated ATM switch, network traffic can be smoothed by a leaky bucket at the output controller for each connection. With the appropriate setting on the leaky bucket parameter, the burstiness of the network traffic can be reduced without increasing the delay in the switch. Therefore, fewer buffers will be required for each active connection within the switch. In this paper, our experimental results have shown that the buffer requirement can be reduced up to 5.75% for each connection, which could be significant, when hundreds of connections are passing through the switches within a regulated ATM network.

In this paper, we introduce a lower bound for the generalized Hamming weights, which is applicable to arbitrary linear code, in terms of the notion of well-behaving. We also show that any [n,k] linear code C over a finite field F is the t-th rank MDS for t such that g(C)+1 t k where g(C) is easily calculated from the basis of Fn so chosen that whose first n-k elements generate C. Finally, we apply our result to Reed-Solomon, Reed-Muller and algebraic geometry codes on Cab, and determine g(C) for each code.

IP multicast is very useful mechanism to deliver data to a large number of receivers such as interactive multimedia sessions. It can not accommodate the heterogeneity of the receivers including network heterogeneity. We propose a multicast traffic controller(s) in a router to solve such situation. A traffic controller has a filter to moderate the output data rate to a link. It makes use of Time-to-Live (TTL) threshold to specify the minimum requirement of a packet. Multimedia data are encoded into multiple layers; basic layer and enhanced layers. By associating TTLs of data layers and the threshold of the filter, we can moderate the traffic by dropping the data of less significant layer. The threshold is dynamically modified according to the local network traffic and link traffic. Our scheme also helps a network and a link(s) avoid from congestion and accommodate other types of traffic at the same time.

Packet contention is one of the fundamental problems that must be overcome in designing packet switches. In banyan network, which has multistage interconnection structure of many small switch elements, we must be concerned with output port conflicts and internal collisions. Dilated banyan network which provides multiple path for internal link can reduce packet loss due to internal collisions in loss system. However, under hot-spot traffic higher packet loss probability is measured at the hot-spot port and the ports close to the hot-spot as coefficient h increases due to the heavy traffic to hot-spot port. In order to moderate the packet loss probability at the hot-spot port we propose the method to disperse the packets which concentrate on the hot-spot route by altering address field of a half of incoming packets. These packets are switched along detour routes. Thus, the traffic concentration toward hot-spot is mitigated and the packet loss probability at the hot-spot port is moderated.

This paper proposes a traffic-adaptive dynamic routing method, which we have named RAG, for connectionless packet networks. Conventional traffic control methods discard the packets which cause congestion. Furthermore, conventional routing methods propagate control messages all over the network for gathering global topology information, and this causes more congestion. In contrast, RAG estimates traffic conditions all over a network without any communication between nodes and makes the best use of free links so that packets make detours to avoid congestive sites. RAG adopts distributed control based on game theory (non-communication, non-zero-sum, two-person). With RAG, nodes play a packet-forwarding game without any communication with each other, and each node controls ordering and routing of the forwarding packets based on the node's individual payoff table which is dynamically reconstructed by observation of surrounding nodes. Nodes cooperate with each other, except for punishment for disloyalty. Repetition of these local operations in nodes aims at the emergence of the gradual network-global traffic balancing. The results of experiments in comparison with the conventional shortest path first (SPF) routing method show that the throughput is about 1.58 times higher with the new method.

Today, many kinds of real-time applications are available over IP networks. It is important to measure the network performance for such applications before making use of real applications. Authors developed the general purpose traffic measurement tool for IP networks. This system can generate any kinds of traffic flexibly and calculate the network performance such as throughput, delay and loss ratio according to received packets. In this paper, the concept of this traffic measurement tool is described in detail, and several examples of network measurements are shown.

The effect of lot size change and test processing logistics on VLSI manufacturing final test process efficiency and cost due to the transition of from conventional 5 or 6 inches to 300 mm (12 inches) in wafer size is evaluated through simulation analysis. Simulated results show that a high test efficiency and a low test cost are maintained regardless of arrival lot size in the range of the number of 300 mm wafers per lot from 1 to 25 and the content of express lots in the range of up to 50% by using WEIGHT+RPM rule and the right final test processing logistics. WEIGHT+RPM rule is the rule that considers the jig and temperature exchanging time, the lot waiting time in queue and also the remaining processing time of the machine in use. The logistics has a small processing and moving lot size equal to the batch size of testing equipment.

This paper proposes a new model which can approximate the delay time distribution in the Internet. It is well known that the delay time in communication links follows the exponential distribution. However, the earlier models cannot explain the distribution when a communication link is heavily overloaded. This paper proposes to use the M/M/S(m) model for the Internet. We have applied our model to the measurement results. This paper deals with one-way delay because it reflects the actual characteristics of communication links. Most measurement statistics in the Internet have been based on round-trip time delay between two end nodes. These characteristics are easily measured by sending sample packets from one node to the other. The receiver side echoes back the packets. However, the results are not always useful. A long distance communication link, such as a leased line, has two different fibers or wires for each direction: an incoming link, and an outgoing link. When the link is overloaded, the traffic in each link is quite different. The measurement of one-way delay is especially important for multimedia communications, because audio and video transmissions are essentially one-way traffic.

This paper proposes a novel sequential coherent preambleless demodulator that uses phase signals instead of complex signals in the automatic frequency control (AFC) and carrier recovery circuits. The proposed demodulator employs a phase-combined frequency error detection circuit and dual loop AFC circuit to achieve fast frequency acquisition and low frequency jitter. It also adopts an open loop carrier recovery scheme with a sample hold circuit after the carrier filter to ensure carrier signal stability within a packet. It is shown that the frame error rate performance of the proposed demodulator is superior, by 30%, to that offered by differential detection in a frequency selective Rayleigh fading channel. The hardware size of the proposed demodulator is about only 1/10 that of a conventional coherent demodulator employing complex signals.

In this paper, we propose an efficient and stable algorithm employing fuzzy logic for available bit rate (ABR) traffic. Large time-delay incurred in the feedback path and the statistical variation in the link capacity at the ATM node can cause instability or unfairness. The stability and fairness issues are discussed and the performance of the proposed mechanism is evaluated.

This paper proposes a medium access control (MAC) protocol for single-hop WDM star-coupler networks, in which the number of stations is larger than the number of channels and the stations have arbitrary distances to the star coupler. The proposed protocol has one control channel for reserving the slots of data channels and several data channels which are used to transmit traffic. This paper also suggests a scheme that accomplishes load balancing among data channels for arbitrary traffic patterns between stations. Since this load balancing scheme diminishes an influence that traffic patterns have on the performance of the proposed MAC protocol, the proposed system is appropriate for systems which have asymmetric traffic intensity between stations. Throughput and mean message delay of the MAC protocol are analyzed using a discrete time Markov process and a D/G/1 queue with batch arrivals. The numerical results show that the performance is improved as the message length increases, the maximum round-trip propagation delay decreases, and the number of data channels increases.

This paper proposes a new high-speed network architecture called Dynamic Transfer Mode, DTM. At the entrance of the DTM network, destination addresses such as IP addresses are converted into DTM routing information and attached to the packet header. In a DTM network, a connection is set up on-the-fly by sending a series of routing link identifiers to the destination, so burst data transfers like WWW traffic are efficiently carried. A connection between adjacent nodes is created and released dynamically within the burst transfer period. This yields higher statistical multiplexing gain and improved bandwidth efficiency compared to with conventional STM. Time division multiplexing is utilized so delay jitter or cell loss, the major drawbacks of Asynchronous Transfer Mode, are avoided. This paper analyzes the performance of a DTM network and describes an implemented switching system. Because a DTM network uses source-routing and passive STM switching, it simplifies the core transit switch while localizing intelligence to edge nodes. A simplified core transit switch is well suited for future high-speed backbone networks.