A timing synchronization is required in the mobile station to determine the correct transmission timing of the mobile-to-base bursts. In this letter, a timing synchronization technique using the reliability check and smoothing algorithm is proposed for the GSM receiver. The reliability check scheme extends the usefulness of this algorithm into low SNR region. And also smoothing algorithm is carried out by a first-order filter with an asymmetric step size. Simulation results show that the proposed algorithm is adequate for timing recovery of GSM modem.

In this paper, an adaptive method for active control of vibration intensity in the frequency domain is proposed. In this method, vibration intensity is observed with the 4-sensor method, and the coefficients of an adaptive FIR filter for the active control is renewed with the Block Filtered-X LMS algorithm in the frequency domain. An experiment with the proposed method is performed on a simple model. As a result, the proposed method gives larger attenuation of vibration intensity than the conventional method in the high frequency region. The overall attenuation in vibration intensity in that frequency region is 14.1 dB with the proposed method, while it is 7.0 dB with the conventional method. In the lower frequency region, the reduction in vibration intensity by the proposed method is roughly equivalent to that obtained by the conventional method. An improvement may also be achieved there by setting the intervals between error sensors properly.

Recently, Miura introduced a construction method of one-point algebraic geometry codes on singular curves, which is regarded as a generalization of one on nonsingular curves, and enables us to construct codes on wider class of algebraic curves. However, it is still not clear whether there really exist singular curves on which we can construct good codes that are never obtained from nonsingular curves. In this paper, we show that for fixed designed minimum distance in a wide range, the dimension of codes on a singular curve is smaller than or equal to that of the codes on its normalization, and the number of check symbols of the former codes is larger than that of the latter codes. This implies the optimality of nonsingular curves for code construction.

A model of the prefrontal cortical circuit has been constructed to investigate the dynamics for working memory processing. The model circuit is multi-layered and consists of a number of circuit modules or columns, each of which has local, excitatory and inhibitory connections as well as feedback connections. The columns interact with each other via the long-range horizontal connections. Besides these intrinsic connections, the pyramidal and spiny cells in the superficial layers receive the specific cue-related input and all the cortical neurons receive a hypothetical bias input. The model cortical circuit amplifies the response to the transient, cue-related input. The dynamics of the circuit evolves autonomously after the termination of the input. As a result, the circuit reaches in several hundred milliseconds an equilibrium state, in which the neurons exhibit graded-level, sustained activity. The sustained activity varies gradually with the cue direction, thus forming memory fields. In the formation of the memory fields, the feedback connections, the horizontal connections, and the bias input all play important roles. Varying the level of the bias input dramatically changes the dynamics of the model cortical neurons. The computer simulations show that there is an optimum level of the input for the formation of well-defined memory fields during the delay period.

The conventional linear prediction can be viewed as a constrained blind equalization problem that has gained a lot of interests along with development of telecommunication networks. Because the blind equalization or deconvolution is a general framework of the inverse problem, the reliable and faster algorithm is requested in many applications. This paper proposes an orthogonal wavelet transform domain realization of a blind equalization technique termed as EVA, and presents an application to speech analysis. An orthogonal transformation has no influence to the equalization result in general, but we show that a particular wavelet makes the matrix in EVA nearly lower triangular that promotes the faster convergence in the estimation of maximum eigenvalue and its associate vector in EVA iteration. The experiments with the Japanese vowels show that the the proposed method effectively separates the glottis and vocal tract information, hence is promising for speech analysis.

A novel method to enhance the practical security of interferometric quantum cryptography is proposed, giving the protocol and detailed constructions including a controlled spontaneous photon emitter, a superradiance amplifier, beam splitters, phase shifters, and a pair of Mach-Zehnder interferometers. The intrinsic uncertainty due to the random phase selection out of three, leads to the detection of eavesdropping. The physical uncertainty of the controlled spontaneous emission of coherent photons also adds temporal equivocation to confuse eavesdroppers.

The optimistic consistency scheme has been established with respect to data consistency and availability in distributed systems. The nomadic data consistency model using version vectors to support data versioning for data synchronization and concurrent conflict detection is suitable for an optimistic replication system that supports large-scale wireless networks. This paper describes the architecture and its data consistency model using data versioning and its access domain control targeted for nomadic data sharing systems, such as collaborative works using database and messaging, and the data transfer optimizations of the model. We evaluate our data versioning scheme comparing with a traditional data versioning and the data transfer optimization by estimation and measurement assuming a mobile worker's job. We generate arithmetic formulas for data transfer estimation using the optimizing techniques and apply them to large-scale data sharing configurations in which collaboration groups are dynamically formed and data is exchanged in each group. The data versioning with an access domain increases flexibility in data sharing configurations, such as mobile collaboration systems and client/server type mobile systems. We confirmed that the combination of the general optimizations and the access domain configurations based on our data consistency model is applicable for large-scale mobile data sharing systems.

As network services become more diverse and powerful, service applications that perform such services are acquiring an ever-larger amount of complicated and changeable relationships. We present a network dependence graph (NDG) that captures both data and control flow relationships between components of service applications that work collaboratively. This graph is constructed based on analysis of both the behavior of each of the service applications and their configuration, which describes the device names they refer to, and allows network slicing to be implemented as a simple graph traversal. Network slicing is the extraction of necessary and minimum service components that may affect the execution of a specified service application; it helps a network manager to find the location of service faults lurking somewhere in the network. We also present a method for locating faults that uses network slicing and a system based on this method.

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.

Auctions are a critical element of the electronic commerce infrastructure. But for real-time applications, auctions are a potential problem - they can cause significant time delays. Thus, for most real-time applications, sealed-bid auctions are recommended. But how do we handle tie-breaking in sealed-bid auctions? This paper analyzes the use of multi-round auctions where the winners from an auction round participate in a subsequent tie-breaking second auction round. We perform this analysis over the classical first-price sealed-bid auction that has been modified to provide full anonymity. We analyze the expected number of rounds and optimal values to minimize communication costs.

IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H. 323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typical case, the bandwidth is cut by 40% for eight G. 723.1-encoded voice streams through header overhead reduction and the number of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.

Certain relationships between successive packet loss patterns and packet spacing are described. To observe a successive packet loss pattern, one possible method is to investigate test packets that are generated at certain intervals (e. g. , Poisson interval, constant regular interval). However, successive packet loss strongly depends on the interval generated. If test packets are generated with long intervals, then successive loss pattern cannot be shown. Thus, in such a method, where the packet intervals may sometimes be long or short, the successiveness of the packet loss should be considered in terms of the packet spacing. To clarify the relationship between the successive packet loss and the packet spacing, we analyze data based on observation of an actual network with the loss window size as a parameter. We find that when the packet spacing is narrower, i. e. , has a shorter interval, the probability becomes higher that the packet immediately following a single packet loss would also be lost.

A new simple method for extracting the capacitance coupling coefficients of sub-0.5-µm flash memory cells is proposed. Different from the previously proposed methods, this method is not affected by a dopant profile of source region because a band-to-band tunneling current from the interface between the drain and the substrate is probed. Use of a reference device eliminates the necessity to make assumptions concerning the electron transport mechanism. Comparison with the other methods shows that the proposed method is simple and accurate.

Congestion Control Scheme of TCP/IP protocol suite is established by Transmission Control Protocol (TCP). Using the self-clocking scheme, TCP is able to maintain a quick optimum connection status for the network path, unless it is given an excessive load to carry to the network. However, in wide area networks, there are some obstructive factors for the self-clocking scheme of TCP. In this paper, we describe the obstructive factors for the self-clocking scheme. We propose a new congestion control scheme using a packet pair scheme and a traffic-shaping scheme. In combining these schemes with TCP, new TCP options and a modification for TCP congestion control algorithms are added. Using our scheme, TCP is able to maintain smooth self-clocking. We implemented this scheme on a network simulator for evaluation. Compared with normal TCP, this scheme was demonstrated to be over 20% more efficient in symmetric communication and over 40% more efficient in asymmetric communication.

In order that the speech recognition system might have a high performance in the noisy environment, the directional microphone arrays at the input of the system have been broadly investigated. The purpose of this study is to develop a new wide-band directional microphone system in view of advancing to an adaptive one afterwards. In the proposed system, three microphones are arranged on a straight line and the beamforming is accomplished in such a way that the output value of the middle microphone is added to the integrated value of the difference between two microphones at both sides. In this study, the signal processing of microphone outputs is implemented by using active RC circuits. Finally, the objective directivity can be experimentally obtained in wide frequency ranges required for the speech recognition.

In this paper we present a complete methodology for efficient electro-mechanical characterization of a CMOS compatible MEMS technology. Using an original test structure, the so-called "U-shape cantilever beam," we are able to determine all mechanical characteristics of force sensors constituted with elementary beams in a given technology. A complete set of electro-mechanical relations for the design of Microsystems have also been developed.

The change over from IPv4 to IPv6 entails a potential increase in the number of records that the Registry System must maintain. Currently, only a few Network Information Centers (NICs), controlled by Internet Assigned Number Authority (IANA), operate their Registry Systems. As they concentrates data into several Registry System, it is not scalable. This paper focuses on the scalability issue in a Registry System and Mie Advanced Registry System (MARS) is proposed. Through the collaboration of independent Registry Systems, MARS ensures data consistency as well as making it possible to access data managed by other Registry Systems. A prototype system of MARS is implemented, maintained and managed on the WIDE 6bone. Some lessen from the operation of MARS give also described.

This paper shows the applicability of the integral equation formulation of the measured equation of invariance (IE-MEI) to two-dimensional dielectric scatterers. That is, a relationship between the scattered electric and magnetic fields, which is derived from the new formulation of the IE-MEI, is applicable to lossless dielectric materials as well as perfect electric conductors (PEC). In addition, we show that the IE-MEI does not suffer from internal resonance problems. These two facts are validated by numerical examples for a circular cylinder and a square cylinder illuminated by Transverse Magnetic (TM) plane wave or a TM line source very close to the scatterers. The numerical results calculated by the IE-MEI agree well with the ones by moment methods that employ combined field formulations with exact boundary conditions.

This paper proposes an algorithm for analyzing the reachability property of Petri nets by the use of unfoldings. It is known that analyzing the reachability by using unfoldings requires exponential time and space to the size of unfolding. The algorithm is based on the branch and bound technique, and experimental results show efficiency of the algorithm.

An optical receiver with a saturated electrical amplifier is studied for signal light that is distorted due to the use of a gain-saturated semiconductor optical amplifier or homowavelength crosstalk light. It is shown that less penalty is induced in a receiver with a DC-coupled saturated amplifier than in one with a linear amplifier, in a practical situation where the decision threshold is fixed at a value optimized for a back-to-back signal. The result suggests that a receiver with a saturated amplifier or a limitter is preferable to an automatic gain control circuit for detecting distorted signal lights.