In this paper, a multiuser receiver using a Hopfield network (Hopfield network receiver) for asynchronous codedivision multiple-access systems is proposed. We derive a novel likelihood function for the optimum demodulation of a data subsequence whose length is far shorter than that of the entire transmitted data sequence. It is shown that a novel Hopfield network receiver can be derived by exploiting the likelihood function, and the derived receiver leads to a low complexity receiver. The structure of the proposed receiver consists of a bank of correlators and a Hopfield network where the number of units is proportional to both the number of users and the length of a data sequence demodulated at a time. Computer simulation results are presented to compare the performance of the proposed receiver with those of the conventional multiuser detectors. It is shown that the proposed receiver significantly outperforms the correlation receiver, decorrelating detector and multistage detector, and provides suboptimum performnace.

This paper describes how to effectively use discriminative training based on Minimum Classification Error (MCE) criterion for a small amount of data in order to attain the highest level of recognition performance. This method is a combination of MCE training and Vector-Field-Smoothed Bayesian learning called MAP/VFS, which combines maximum a posteriori (MAP) estimation with Vector Field Smoothing (VFS). In the proposed method, MAP/VFS can significantly enhance MCE training in the robustness of acoustic modeling. In model training, MCE training is performed using the MAP/VFS-trained model as an initial model. The same data are used in both trainings. For speaker adaptation using several dozen training words, the proposed method has been experimentally proven to be very effective. For 50-word training data, recognition errors are drastically reduced by 47% compared with 16.5% when using only MCE. This high rate, in which 39% is due to MAP, an additional 4% is due to VFS, and a further improvement of 4% is due to MCE, can be attained by enhancing MCE training capability by MAP/VFS.

In this paper, the bit error rate (BER) performance of the Spread Spectrum communication system with Constrained Spreading Codes (SS-CSC) is analyzed. The BER of the SS-CSC system is the same as that of the Bi-orthogonal system. Moreover, the frequency utilization efficiency of the SS-CSC system is better than that of the Bi-orthogonal system when K 10 and N = 3.

This paper describes new problems in delay analysis for high-performance LSI design and presents a static delay analysis tool PCHECK. PCHECK is characterized by (1) a new critical path trace algorithm for avoiding the error caused by signal transient time and (2) a precise delay calculation model for resistive shielding. Experimental results show that the delay calculation error in the worst case is less than 20 ps.

The pros and cons of CDMA as a multiple access scheme for third generation cellular mobile radio systems are considered. Main criteria are spectral efficiency and capacity, but also flexibility and costs.

This paper examines the performance of a direct sequence, spread spectrum (DSSS) multiple access (MA) system used over two typical, frequency-selective, fading satellite channels. In an attempt to increase the system efficiency, an adaptive receiver described by Rapajic and Vucetic [1] has been implemented. This system has been combined with soft-decision convolutional coding in order to improve the system performance under the fading conditions relative to the uncoded system and to allow as many simultaneous users as possible. Various code rates have been examined and the results are given. This paper specifically focuses on DSSS-MA systems with low spreading ratios. The satellite channels used in this paper were produced by models developed as a result of experimental measurements of fading satellite channels for rural and urban environments.

In order to highlight a rapid progress attained in the field of millimeter waves in Japan, this paper describes several key topics including transistors, integrated circuits, planar antennas, millimeter wave photonics, and others.

When wireless multi-media information which includes speech, image, data and so on are transmitted, the defference in information rate, required quality as well as traffic performance should be taken into account. A wireless spread spectrum system can achieve a flexible balance of these differences because of the inherent asynchronous capability of CDMA. In this paper, we propose a wireless multi-media CDMA system based on a processing gain control in a dynamic traffic channel. According to the priority of each medium and channel measurement information i.e. traffic, the optimal processing gain can be controlled by using Nonlinear Programming. Numerical results demonstrate that the proposed method possesses higher flexible capacity than TDMA in a dynamic multi-medea traffic channel.

A digital filter is one of the fundamental elements in the digital video transmission, and a multiplier acts as the key factor that determines the operation speed and silicon area of the filter. In terms of the digital video transmission, the required performance of a multiplier is to operate at the speed of 20-100 MHz but with the precision of 8-10 bits. In the case of implementing such an FIR filter with more than a certain number of taps, the same number of multipliers are necessary to realize the speed. Moreover, even though the coefficients to the filter are desired to be programmable, it is possible to change coefficients in the vertical fly-back interval of television receivers. This allows the preloadability of coefficients to the filter such that each coefficient can be treated as a constant during the filtering operation. Motivated by these requirements and functionalities, a novel multiplier and FIR filter architecture is described, which is to be synthesized with the use of a high level synthesis tool of COMPASS Design Navigator, partly with the aid of the manual design by means of a 0.8µm CMOS library.

A generalized Reed-Muller expression (GRM) is obtained by negating some of the literals in a positive polarity Reed-Muller expression (PPRM). There are at most 2(n2)^(n-1) different GRMs for an n-variable function. A minimum GRM is one with the fewest products. This paper presents certain properties and an exact minimization algorithm for GRMs. The minimization algorithm uses binary decision diagrams. Up to five variables, all the representative functions of NP-equivalence classes were generated and minimized. Tables compare the number of products necessary to represent four-and five-variable functions for four classes of expressions: PPRMs, FPRMs, GRMs and SOPs. GRMs require, on the average, fewer products than sum-of-products expressions (SOPs), and have easily testable realizations.

Transport-processing FPGAs have been proposed for flexible telecommunication systems. Since those FPGAs have finer granularity of logic functions to implement circuits on them, the amount of routing resources tends to increase. In order to keep routing congstion small, it is necessary to execute placement and routing simultaneously. This paper proposes a simultaneous placement and global routing algorithm for transport-processing FPGAs whose primary objective is minimizing routing congestion. The algorithm is based on hierarchical bipartition of layout regions and sets of LUTs (Look Up Tables) to be placed. It achieves bipartitioning which leads to small routing congestion by applying a network flow technique to it and computing a maximum flow and a minimum cut. If there exist connections between bipartitioned LUT sets, pairs of pseudo-terminals are introduced to preserve the connections. A sequence of pseudo-terminals represents a global route of each net. As a result, both placement of LUTs and global routing are determined when hierarchical bipartitioning procedures are finished. The proposed algorithm has been implemented and applied to practical transport-processing circuits. The experimental results demonstrate that it decreases routing congestion by an average of 37% compared with a conventional algorithm and achieves 100% routing for the circuits for which the conventional algorithm causes unrouted nets.

For quasi millimeter-wave and millimeter-wave high-speed wireless communications over wireless LANs and wireless ATMs, narrow beam antennas have been shown to provide high transmission quality by suppressing the troublesome multipath effect. However, the diameter of sector antennas needed to create the narrow beams rapidly increases with the sector number. In addition, the cylindrical shape of typical sector antennas does not suit portable terminals. This paper shows a methodology for designing planar sector antennas that overcomes these problems. The proposed antenna uses two kinds of beams and the antenna gains are equalized in all sectors. The antenna is developed as a 4-beam subarray fed by a planar Butler matrix circuit. The design method of the subarray and an evaluation of its characteristics in the 20 GHz band are discussed.

In the envisaged Universal Mobile Telecommunications System (UMTS), the satellite component will have to provide services to mobile or, in some cases, hand held terminals with a required grade of user co-operation and link availability in various communication environments. This may require the capability of the satellite link to cope with more severe multipath environments than those for which mobile satellite links are most frequently designed (maritime or open rural applications); unfortunately, when the mobile radio channel is affected by multipath and a coherent demodulation is chosen, the phase synchronisation can be a critical issue. To satisfactorily deal with the arising difficulties, a dual channel demodulation is a viable and efficient strategy for the forward link, since only one common pilot channel is needed in this case. If the same dual channel demodulation is considered for the return link, an unacceptable capacity reduction may result. In this paper, some synchronisation strategies are analysed and an efficient dual channel demodulation scheme is proposed for the return link of a satellite DS-CDMA mobile communication system; furthermore, the impact on the overall system performance or capacity is analysed.

In this paper, a simple frame synchronization system for M-ary/SS communication systems is proposed, and synchronization performance and the resulting bit error rate performance are analyzed. The frame synchronization system uses racing counters and framing chips which are added to spreading sequences. M-ary/SS communication systems can improve bit error rate performance under the condition in which there is an additive white gaussian noise. Synchronization of M-ary/SS communication systems is difficult, however, because M-ary/SS communication systems have several spreading sequences. The authors proposed the simple frame synchronization system which uses only one chip in the spreading sequence as a framing signal. This system needs a long time for initial acquisition as the frame length is longer. The proposed system in this paper can make initial acquisition time short by increasing the number of framing chips. The proposed system corresponds to the conventional system when the number of framing chips is l. As the result, it is shown that several framing chips contribute to decrease the initial acquisition time. Moreover, the frame synchronization system can be applied to asynchronous M-ary/SSMA system when different framing chip pattern is assigned to each user.

The Time-Slicing paradigm is a newly developed method for the training of neural networks for speech recognition. The neural net is trained to spot the syllables in a continuous stream of speech. It generates a transcription of the utterance, be it a word, a phrase, etc. Combined with a simple error recovery method the desired units (words or phrases) can be retrieved. This paradigm uses a recurrent neural network trained in a modular fashion with natural connectionist glue. It processes the input signal sequentially regardless of the input's length and immediately extracts the syllables spotted in the speech stream. As an example, this character string is then compared to a set of possible words, picking out the five closest candidates. In this paper we describe the time-slicing paradigm and the training of the recurrent neural network together with details about the training samples. It also introduces the concept of natural connectionist glue and the recurrent neural network's architecture used for this purpose. Additionally we explain the errors found in the output and the process to reduce them and recover the correct words. The recognition rates of the network and the recovery rates for the words are also shown. The presented examples and recognition rates demonstrate the potential of the time-slicing method for continuous speech recognition.

Recently , a short range millimeter wave or a microwave sensing system has been extensively studied to estimate a target position or a source location. It can be applied to indoor propagation analysis, carborne applications, etc. The application of the superresolution technique has been proposed to obtain a high resolution performance in the time domain or the spatial domain. However, the availability of the polarization synthesis in the receiving antennas has not been considered. In this paper, we use a pair of polarized swept frequency data and propose two modifications of the MUSIC algorithm to enhance the resolution of time delay. One modification is the correlation matrix formulation which relates to the total signal power, and the other is a polarization filtering applied to the correlation matrix. These modifications have advantages such that. 1)Reduction of the estimation problem to the delay time estimation only; 2)Easy implementation. Experimental results are illustrated to show the availability of the methods, and to confirm the high resolution performance compared with the conventional method.

This paper gives an overview of the research and development trends in millimeter-wave short-range application systems, such as communication systems and sensing systems, in Japan and other countries. Frequency management trends are also described. Major research and development efforts in Japan have currently been concentrated on the 59-64 GHz band. The first major achievement resulting from those efforts was the allocation of the 60-61 GHz band to the automotive radar systems. Test productions of automotive radars in this band have already started. Further technological developments to reduce the cost and size of radar products are, however, required in order for such radar systems to be widely used. Development of broadband wireless LAN systems has also been intensively made in the 60 GHz band. In addition, technical issues related to standardization of millimeter-wave wireless LAN systems in the 60 GHz band have been examined at the Association of Radio Industries and Businesses. The application areas of millimeter-waves in the future are expected to become more diverse. Research and development trends of future application systems, such as broadband mobile communication systems and imaging radar systems, are also described. These systems require more advanced millimeter-wave technologies, such as smart antennas, low power-consumption devices, and more sensitive detectors. Efforts to develop these technologies must be strengthened.

The inclusion of the non-quasi-static effect is crucial in the simulation of the microwave circuits for MOS transistors. This report proposes a simple model which includes this effect in small-signal simulation. The simulated results are consistent with the measured data up to a frequency that is 30 times higher frequency than the cut-off frequency.

Two simple and high performance multichannel distributed queue dual bus (MDQDB) protocols based on the wavelength division multiplexing (WDM) network technology are proposed, and the network architecture and operations are presented. To be suited for a high speed network, they inherit the advantages of the DQDB protocol such as node simplicity, network flexibility and distributed operations of individual nodes. The network capacity can also be greatly increased with marginal increase of node complexity. Simulation has been done to estimate the performances such as throughput and average access delays for individual nodes. The influence of the bandwidth balancing mechanism on the two protocols is considered at medium, high, and overload conditions. We also investigate the fairness characteristics in asymptotic conditions for various initial states.

This paper deals with the synthesis of Petri nets. Partial languages adequately represent the concurrent behaviors of Petri nets. We first propose a construction problem for Petri nets, in which the objective is to synthesize a Petri net to exhibit the desired behavior specified as a partial language. We next discuss the solvability of this problem and last present the cutline of a solution technique.