The problem of finding a path with two additive constraints, in particular finding a path that satisfies both the cost and the delay constraints, is called multi-constrained path (MCP) problem in the literature. In this paper, we explore the MCP problem based on the idea of single mixed weight --a mixed weight for each link is first obtained by combining its delay and cost, and then Dijkstra's algorithm is used to find the corresponding shortest path. Given two infeasible paths, it can be theoretically proved that a better path can possibly be found if we choose an appropriate parameter to construct the mixed weight. An approximate algorithm is thus proposed to solve the MCP problem. Theoretical analysis demonstrates that this algorithm can make a correct judgment whether there is a feasible path or not with a very high probability even in the strict case where the delay bound is between the delays of the least delay path and the least cost path, while the cost bound is between the costs of the two paths. On the other hand, the time complexity of this algorithm is very small since it only needs to execute Dijkstra's algorithm a limited number of times. The excellent performance of the proposed algorithm is verified by a large number of experiments on networks of different sizes.

This paper proposes a novel space diversity reception scheme suitable for packet-based orthogonal frequency division multiplexing (OFDM) wireless access systems that achieves large diversity gain by improving the accuracy of both carrier frequency synchronization and phase tracking. Phase tracking compensates the phase rotation caused by residual carrier frequency error and phase noise and is necessary for high data rate OFDM systems that use coherent detection. In the proposed scheme, the accuracy of carrier frequency synchronization is improved by combining the information of the carrier frequency offset detected on all diversity branches; the accuracy of phase tracking is improved by using pilot signals whose signal to noise ratio (SNR) is raised by maximal ratio combining of the pilot signals extracted from all branches. Computer simulation results show that the proposed scheme effectively improves the diversity gain even in severe environments such as those with low carrier to noise ratios (CNR) and large delay spreads.

In optical networks, wavelength converters are required to improve the efficiency of wavelength-division multiplexing. In this paper, we propose a genetic algorithm to determine the optimal locations of the nodes in the network where a given number of converters are placed. Optimality is achieved by the minimum wavelength blocking probability. Our algorithm is applied to two realistic networks constructed from the locations of major cities in Ibaraki Prefecture and from those in Kanto District in Japan and is shown to reach the nearly optimal solution in a limited number of generations. The accuracy is verified by simulation. The computational time is compared with that of an exhaustive search algorithm.

We propose a new type of microstrip line to waveguide transition fabricated on a single layer dielectric substrate. Impedance matching of the transition is achieved by controlling the size of a matching element and the length of an inserted microstrip line across a waveguide. As a result of experiments, low transmission loss of 0.4 dB is realized at the design frequency of 76.5 GHz. Bandwidth of the transition is numerically investigated by the finite element method. It is clarified that the bandwidth of the transition becomes wider as the cross section of the waveguide becomes smaller and twice as wide as that of a conventional microstrip patch antenna element fabricated on a dielectric substrate with the same parameters. In addition, the effect of errors in relative position between the dielectric substrate and the waveguide is also investigated. It becomes clear that degradation of transmission characteristics is caused by the shift of resonant frequency and keeps less than 0.1 dB for a manufacturing accuracy within 0.1 mm.

With the progress of integration of circuits and PCBs (Printed Circuit Boards), novel techniques have been required for verification of signal integrity. Noise analysis of the power/ground planes is one of the most important issues. This paper describes a high-speed simulator for PCBs which contain the interconnects with nonlinear terminations. This simulator is based on the environmental tool ASSIST (Assistant System for Simulation Study) constructed for development of the circuit simulators, and is combined with PRIMA (Passive Reduced-Order Interconnect Macromodeling Algorithm). In this simulator, an efficient implementation of PRIMA is considered with using a voltage-controlled current source (VCCS) model. Finally, this simulator is applied to the analysis of power/ground planes of the simple PCBs, and the validity is verified.

This paper proposes design of new linear bipolar OTAs using hyperbolic circuits with an intermediate voltage terminal. Four types of the OTAs are presented; two OTAs contain a hyperbolic sine circuit and the other two OTAs employ a hyperbolic cosine circuit. The linear input voltage range of the proposed OTAs is wider than that of the well-known conventional OTA, multi-TANH doublet, while each proposed OTA has advantages, such as low power dissipation, high-frequency characteristics and so on. The results of SPICE simulation show that satisfactory characteristics are obtained.

Many researches have been conducted on the recognition of Thai characters. Different approaches, such as neural network, syntactic, and structural methods, have been proposed. However, the success in recognizing Thai characters is still limited, compared to English characters. This paper proposes an approach to recognize the printed Thai characters using the hybrid of global feature, local features, fuzzy membership function and the neural network. The global feature classifies all characters into seven main groups. Then the local features and the neural network are applied to identify the characters.

This paper deals with two-processor nonpreemptive scheduling problem for acyclic SWITCH-less program nets including two types of nodes: AND-node and OR-node. Compared with task graphs that are a special case of acyclic SWITCH-less program nets and include only AND-nodes, the multiprocessor scheduling problem of general acyclic SWITCH-less program nets is more complicated. Since multiprocessor scheduling problem for general task graphs is NP-hard, so is for acyclic SWITCH-less program nets generally. In this paper, we suppose the acyclic SWITCH-less program nets satisfy: (i) each AND-node and OR-node have 1 and 0 node firing time, respectively; (ii) each AND-node possesses single input edge. For such a class of acyclic SWITCH-less program nets, we first propose a hybrid priority list L that consists of both dynamic and static sub-lists. Then we prove that, for a given acyclic SWITCH-less program net to be executed by two identical processors, the schedule generated by L is optimal.

The current paper proposes a new method for reducing the computational overhead of fine-to-coarse multi-resolution motion estimation (MRME) on the finest resolution level based on limiting the search region used to consider the motion vectors from the coarsest resolution subband. Half-pel accuracy motion estimation (HPAME) is used in the baseband to achieve an exact motion vector, which has a significant influence on the reconstructed image. However, since this method involves a high computational overhead, the use of selective interpolation is suggested to reduce the computational overhead. The proposed method uses HPAME to estimate exact motion vectors in the baseband, then based on the characteristics of these motion vectors, the motion vectors in the higher frequency subbands are selectively estimated. That is, motion vectors in the higher frequency subbands are only estimated for those blocks with half-pixel accuracy motion vectors in the baseband. Experimental results show that the proposed algorithm can produce better image quality results than the conventional algorithms.

This paper presents a new DSP-oriented code optimization method to enhance performance by exploiting the specific architectural features of digital signal processors. In the proposed method, a source code is translated into the static single assignment form while preserving the high-level information related to loops and the address computation of array accesses. The information is used in generating hardware loop instructions and parallel instructions provided by most digital signal processors. In addition to the conventional control-data flow graph, a new graph is employed to make it easy to find auto-modification addressing modes efficiently. Experimental results on benchmark programs show that the proposed method is effective in improving performance.

We consider the services of broadcasting short messages via forward signaling channels in wireless cellular networks. In the provision of such services, the negative effect of short messages on the delivery of delay-sensitive control messages must be restricted. On the other hand, it is desirable to accommodate the users' demands for service enhancements involving timeliness and informativeness. As a way to resolve such conflicting arguments, we present a generic scheme in which a short message may be rejected or removed according to the buffer occupancy at the base station and is split into a number of segments for the transmission across a forward signaling channel. However, the rejection, removal and segmentation exhibit a trade-off among several facets of service enhancements. Thus, for a quantitative evaluation of the scheme and efficient optimization of design parameters, we develop an analytical method to calculate the moments of delay times experienced by control and short messages at a base station. Using the analytical method, we investigate the delay and loss performance of control and short messages with respect to the message load and short message length.

This paper presents a refined model for the fuzzy logic implementation of an available bit rate (ABR) flow control switch. This refined model is named fuzzy explicit rate (FUER) switch mechanism. FUER switch mechanism is designed to effectively perform congestion control on ABR traffic in asynchronous transfer mode (ATM) networks. The performance of FUER scheme is evaluated against those of two other explicit rate (ER)-based switch mechanisms using simulation in particular local area network (LAN) and wide area network (WAN) environments. On the whole, FUER scheme performs better than the other schemes. Although it has the smallest control parameter set, it is a more efficient and scalable ER-based switch mechanism.

The IETF Mobile IP defines two multicast options: remote subscription (RS) and bi-direction tunneling (BT). In order to synthesize the strong points of these two IETF multicast options, we propose a hybrid approach, mMOM, which selectively uses two IETF multicast options based on the mobility of mobile hosts. Whenever a mobile host requests its first registration to a certain foreign agent, the corresponding foreign agent starts the service using the BT option. Afterwards, if it requests re-registration to the same foreign agent, the foreign agent considers it to be relatively immobile and continues services using the RS option. We propose a new metric to compare heterogeneous algorithms. Simulation results show that our approach outperforms all others.

We proposed the conduction mechanism of organic light-emitting diode (OLED) using a one-dimensional discontinuous model. We assumed that each emitting molecule corresponds to a hopping site according to the actual charge transfer between adjacent molecules. Both carrier mobility of Alq3 and barrier heights for each carrier were derived from experimental data. We calculate transient behavior of carrier, field, and exciton distribution. Both carrier injections assumed the Schottky injection. In the previous results, when we assumed that calculated current density fit the experimental one in the current density field curve, calculated light-emission intensity did not fit the experimental one in the light-emission field curve. Furthermore, the slope of the calculated light emission-field curve is too small to fit the experimental one. In the previous study, hopping distance was assumed to be 1 nm. In this study, it is assumed to be 1.7 nm. We consider that field dependence of electron injection is too weak to explain only the Schottky emission. When the electron injection is assumed to be both Schottky emission and Fowler-Nordheim emission calculated light-emission field as well as the current-density field curves were fit to the curve of each experimental characteristics.

This paper describes a method for hands-free speech recognition, and particularly for the simultaneous recognition of multiple sound sources. The method is based on the 3-D Viterbi search, i.e., extended to the 3-D N-best search method enabling the recognition of multiple sound sources. The baseline system integrates two existing technologies--3-D Viterbi search and conventional N-best search--into a complete system. Previously, the first evaluation of the 3-D N-best search-based system showed that new ideas are necessary to develop a system for the simultaneous recognition of multiple sound sources. It found two factors that play important roles in the performance of the system, namely the different likelihood ranges of the sound sources and the direction-based separation of the hypotheses. In order to solve these problems, we implemented a likelihood normalization and a path distance-based clustering technique into the baseline 3-D N-best search-based system. The performance of our system was evaluated through experiments on simulated data for the case of two talkers. The experiments showed significant improvements by implementing the above two techniques. The best results were obtained by implementing the two techniques and using a microphone array composed of 32 channels. More specifically, the Word Accuracy for the two talkers was higher than 80% and the Simultaneous Word Accuracy (where both sources are correctly recognized simultaneously) was higher than 70%, which are very promising results.

A drain avalanche hot carrier lifetime model including a body effect caused by secondary hot electrons has been developed. It has been confirmed that the proposed model fits a wide range of experimental data using a small number of parameters. The model provides a practical modeling methodology for reliability simulation based on parameter extraction at maximum substrate current conditions alone. Simulation accuracy produced by the methodology has been experimentally verified using ring oscillators including NAND gates. It has been demonstrated that simulation accuracy of degradations has become by 0.34 decade better using the new methodology than using that based on the conventional τId/W-Isub/Id model.

A new type of electromagnetic vibration transducer for use in an IME (implantable middle ear) system is presented and evaluated by in-vitro experiment. Because the new designed transducer includes two magnets glued together with the same pole facing inside the coil, it can reduce the interference from an environmental magnetic field. And the proposed transducer exhibits a high vibration efficiency and wide frequency response. Using dead human's temporal bone, in-vitro experiments were carried out and the results showed that the proposed vibration transducer will be properly used to implantable middle ear for mild to severe hearing loss patients.

A new H.263+ rate control method that has very low encoder-decoder delay, small buffer and low computational complexity for hardware realization is proposed in this paper. This method focuses on producing low encoder-decoder delay in order to solve the lip synchronization problem. Low encoder-decoder delay is achieved by improving target bit rate achievement and reducing processing delay. The target bit rate achievement is improved by allocating an optimum frame encoding bits, and employing a new adaptive threshold of zero vector motion estimation. The processing delay is reduced by simplifying quantization parameter computation, applying a new non-zero coefficient distortion measure and utilizing previous frame information in current frame encoding. The simulation results indicate very large number skipped frames reduction in comparison with the test model TMN8. There were 80 skipped frames less than that of TMN8 within a 380 frame sequence during encoding of a very high movement video sequence. The 27 kbps target bit rate is achieved with insignificant difference for various types of video sequences. The simulation results also show that our method successfully allocates encoding bits, maintains small data at the encoder buffer and avoids buffer from overflow and underflow.

The multicast ATM switch has been developed for the purpose of the point-to-multipoint cell transmission. The basic structure of conventional multicast ATM switches is mainly based on a T. T. Lee's multicast switch, since it has a very simple and expandable structure. However, in spite of these benefits, it requires excessive hardware for the loss-free cell transmission, since it employs the blocking network, i.e., the banyan network as a copy network and a routing network. In this paper, we propose a new network for the multicast ATM switching. In proposed copy network, we adopt a new structure, the parallel broadcast banyan network with bypass links between switch planes, to offer the maximum cell transmission capacity and the fault tolerance. All conflict cells, which are blocked during the cell routing process, are bypassed to the next switch plane through bypass links and try to be routed. And to support the highly efficient cell transmission, we propose Alternate Path Scheme (APS) and copy-number (CN) comparator in the proposed copy network. APS is a kind of cell transmission schemes and guarantees multicasting capability to achieve a high performance. To estimate the performance of a proposed copy network, we provide several simulation results.

An LMS adaptive digital filter using distributed arithmetic (DA-ADF) has been proposed. Cowan and others proposed the DA adaptive algorithm with offset binary coding for the simple derivation of an algorithm and the use of an odd-symmetry property of adaptive function space (AFS). However, we indicated that a convergence speed of this DA adaptive algorithm degraded extremely by our computer simulations. To overcome these problems, we have proposed the DA adaptive algorithm generalized with two's complement representation and effective architectures. Our DA-ADF has performances of a high speed, small output latency, a good convergence speed, small-scale hardware and lower power dissipation for higher order, simultaneously. In this paper, we analyze a convergence condition of DA adaptive algorithm that has never been considered theoretically. From this analysis, we indicate that the convergence speed is depended on a distribution of eigenvalues of an auto-correlation matrix of an extended input signal vector . Furthermore, we obtain the eigenvalues theoretically. As a result, we clearly show that our DA-ADF has an advantage of the conventional DA-ADF in the convergence speed.