Orthogonal Frequency Division Multiplexing-Direct Sequence/Code Division Multiple Access (OFDM-DS/CDMA) systems provide frequency diversity gain avoiding inter symbol interference (ISI) in a frequency selective fading channel. However, path diversity gain can not be obtained by using conventional OFDM-DS/CDMA schemes. This paper proposes a new multiple antenna transmission system with combined path diversity and frequency diversity. Signal of each antenna is delayed by several chips to create artificial path diversity as well as frequency diversity of multi-carrier transmission in which can then be combined by using a RAKE receiver. Therefore multiple antenna transmission scheme creates a path diversity effect on uncorrelated signals in multi-carriers from each antenna. The received uncorrelated signals can be processed by Maximum Ratio Combining (MRC) diversity without ISI at a RAKE receiver even when we use FFT modulation. As a result, we can obtain combined path diversity and frequency diversity gain effectively by the RAKE system with the combination of multiple antennas.

We propose middleware which works on widely-used commercial off-the-shelf platforms (UDP/IP, FastEthernet, and Windows NT or commercial real-time kernels) to realize real-time distributed services for plant monitoring and control systems. It is not suitable to use TCP/IP for the systems because of its unpredictable re-transmission, while, as well known, UDP/IP does not guarantee certain arrivals of packets and it is also not acceptable for the systems. With UDP/IP, packets are lost mainly because of collisions in a network and buffer overflows. To avoid these packet losses, the middleware controls scheduling of all the packets transmitted between the nodes in a distributed system and prevents excessive collisions and buffer overflows. The middleware provides a necessary set of functions for plant monitoring and control applications. The middleware on each node in a distributed system consists of library functions and run-time modules. An application program on the node is required to use these library functions according to the rules the middleware provides. In this way the middleware can manage all the traffic among the nodes in the system. Receiving requests from the application via library functions, the run-time module of the middleware schedules transmission of messages to other nodes, avoiding unexpected delivery delays or buffer overflows. The module also guarantees application-to-application quality of service (QoS), such as transmission period and delivery deadline, required by the applications. This is achieved by assigning the resources not shared by other services to each distributed service and scheduling these resources so as not to violate the assignment. Here, resources include maximum numbers of packets which a node can receive or send in a specific period (20 msec, for example). We show implementation of the middleware to make it clear how to guarantee application-to-application QoS with some application examples.

In quantum communication theory, a realization of the optimum quantum receiver that minimizes the error probability is one of fundamental problems. A quantum receiver is described by detection operators. Therefore, it is very important to derive the optimum detection operators for a realization of the optimum quantum receiver. In general, it is difficult to derive the optimum detection operators, except for some simple cases. In addition, even if we could derive the optimum detection operators, it is not trivial what device corresponds to the operators. In this paper, we show a realization method of a quantum receiver which is described by a projection-valued measure (PVM) and apply the method to 3-ary phase-shift-keyed (3PSK) coherent-state signals.

The maximum amounts of embedded information that is important in practical system design of two watermarking methods based on the parallel combinatorial spread spectrum (PC/SS) scheme are discussed in this paper. One is a private watermarking method proposed in this paper and has a practical strong point to make the quality of the watermarked image to be constant in any images. The other is a public watermarking method that was previously proposed by the authors. Through this study, the minimum number of orthogonal sequences for embedding the required amount of information under the condition that quantization noise is only assumed is found in each watermarking method.

We have developed a signal processing method that is appropriate for detecting electromagnetic radiation due to earthquake activities. The radiation is usually accompanied by a background noise that is mainly caused by atmospheric discharges in the tropical regions. Data representing the seismic radiation is presented as sound via the concept of sonification. This is useful for immediately finding out anomalous seismic radiations, which are often followed by a disastrous earthquake, from the massive data collected from over forty observation stations. It is illustrated that the auditory display is valuable for future earthquake prediction systems.

This work presents an analysis of IDDQ dependency on the primary current that flows through the bridging fault and driven gates current. A maximum primary current depends only on the test vectors which minimize channel resistances of transistors. The driven gates current generates when intermediate voltage occurs on the faulty node with creation current path between VDD and GND through the driven gates, and its value depends on circuit parameters such as transistor sizes and fan-in number of driven gates.

Recently many researches concerning heart sound analysis are being processed with development of digital signal processing and electronic components. But there are few researches about recognition of heart sound, especially full cardiac cycled heart sound. In this paper, three new recognition methods about full cardiac cycled heart sound were proposed. The first method recognizes the characteristics of heart sound by integrating important peaks and analyzing statistical variables in time domain. The second method builds a database by principal components analysis on training heart sound set in time domain. This database is used to recognize new input of heart sound. The third method builds the same sort of the database not in time domain but in time-frequency domain. We classify the heart sounds into seven classes such as normal (NO) class, pre-systolic murmur (PS) class, early systolic murmur (ES) class, late systolic murmur (LS) class, early diastolic murmur (ED) class, late diastolic murmur (LD) class and continuous murmur (CM) class. As a result, we could verify that the third method is better efficient to recognize the characteristics of heart sound than the others and also than any precedent research. The recognition rates of the third method are 100% for NO, 80% for PS and ES, 67% for LS, 93 for ED, 80% for LD and 30% for CM.

Off-line state identification methods for a sequential machine using a homing sequence or an adaptive homing sequence (AHS) are well-known in the automata theory. There are, however, so far few studies on the subject of the on-line state estimator such as a state observer (SO) which is used in the linear system theory. In this paper, we shall construct such an SO for a Moore machine based on the state identification process by means of AHSs, and discuss the convergence property of the SO.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

In this letter, we present a method to generate sets of sequences suitable for multicode transmission in quasi-synchronous (QS) CDMA systems. We focus on Gold code but extension to orthogonal Gold code is straightforward. We show that by appropriate classification of sequences, it is possible to have sets whose cross correlation is small in QS situations.

We consider the problem of efficient learning of probabilistic concepts (p-concepts) and more generally stochastic rules in the sense defined by Kearns and Schapire and by Yamanishi. Their models extend the PAC-learning model of Valiant to the learning scenario in which the target concept or function is stochastic rather than deterministic as in Valiant's original model. In this paper, we consider the learnability of stochastic rules with respect to the classic 'Kullback-Leibler divergence' (KL divergence) as well as the quadratic distance as the distance measure between the rules. First, we show that the notion of polynomial time learnability of p-concepts and stochastic rules with fixed range size using the KL divergence is in fact equivalent to the same notion using the quadratic distance, and hence any of the distances considered in [6] and [18]: the quadratic, variation, and Hellinger distances. As a corollary, it follows that a wide range of classes of p-concepts which were shown to be polynomially learnable with respect to the quadratic distance in [6] are also learnable with respect to the KL divergence. The sample and time complexity of algorithms that would be obtained by the above general equivalence, however, are far from optimal. We present a polynomial learning algorithm with reasonable sample and time complexity for the important class of convex linear combinations of stochastic rules. We also develop a simple and versatile technique for obtaining sample complexity bounds for learning classes of stochastic rules with respect to the KL-divergence and quadratic distance, and apply them to produce bounds for the classes of probabilistic finite state acceptors (automata), probabilistic decision lists, and convex linear combinations.

This paper discusses a communication system with a multiple-access channel where two users simultaneously send complex-valued signals in the same frequency-band. In this channel, ambiguity in decoding occurs when receiver trying to estimate each users' signal. In order to solve the ambiguity problem, a family of uniquely decodable code is derived in this paper. The uniquely decodable code is designed by using trellis-coded modulation (TCM) pair where the trellis structure of one TCM is a transformation of the other in the pair. It is theoretically proved that, with the proposed coding scheme, the composite received signal can be uniquely decomposed into the two constituent signals for any power ratio and any phase difference between the received two users' signals. Improvement of BER performance over non-uniquely decodable code is illustrated by computer simulation.

FDMA/TDMA non-geostationary earth orbit satellite systems (Non-GEOS) generally require a pre-planned pool of radio resource, i.e., frequency and time slot plan (FTSP), for each gateway earth station (GES) prior to the real-time channel assignment by the multiple GES's sharing the resources harmoniously. The time-variant nature of those systems implies that a dynamic FTSP planning method is crucial to the operation to cope with the time-variant traffic demand and the inter-beam interference condition. This paper proposes and compares three algorithms (Serial-numbering, DP-based, and Greedy algorithms) mixed with two strategies (concentrated- and spread-types) for the resource allocation. The numerical evaluation demonstrates that Greedy algorithm with the spread-type strategy is very effective and promising with feasible calculation time for the FTSP generation.

In this paper, we propose a generalized frequency assignment algorithm to minimize the intermodulation products caused by nonlinear amplification in satellite transponder. We also analyze the performance of proposed algorithms in terms of C/IM and execution time. Most of the published algorithms are too restrictive to be applied to the frequency planning of many realistic systems that are usually characterized by multi-level and/or multi-bandwidth. In developing the proposed "TDTI algorithm," we utilized and modified basic concepts of Okinaka's DELINS-INSDEL algorithm to extend its applicability from one-level systems to more general systems. We also propose a modified version of TDTI algorithm called "WTDI-SDELINS" to circumvent the problem of relatively long execution time.

This paper presents a new and robust technique for a coarse frequency offset estimation in OFDM system. As an evaluation of the proposed algorithm, we apply it to Eureka 147 DAB system. The proposed coarse frequency offset estimation algorithm is based on the differential detection technique between adjacent subcarriers to eliminate the phase shift effects of symbol timing offset and fractional frequency offset. Coarse frequency offset is determined from the correlation outputs between a received intercarrier differential phase reference symbol and several locally generated but frequency shifted intercarrier differential phase reference symbols. The performance of our estimation algorithm is evaluated by means of computer simulation and compared with that of previous proposed algorithms for DAB transmission modes I, II, III and IV. Simulation results show that the proposed algorithm generates extremely accurate estimates with a low complexity irrespective of the symbol timing offset.

As noncoherent direct sequence code-division multiple-access (DS-CDMA) mobile satellite communications, two typical transmission schemes are compared; one is a quasi-synchronous differential BPSK (QS-DBPSK) where orthogonal signals are used for reducing the multiple access interference and the other is M-ary orthogonal signaling (MOS) scheme where orthogonal signals are used for exploiting more efficient modulation. The performances are evaluated in additive white Gaussian noise (AWGN) and shadowed Rician fading (SRF) channels and the effects of timing misalignments in the QS-DBPSK system and the amount of Doppler shifts of a SRF channel are investigated. The results show that MOS much outperforms QS-DBPSK in the region of low system loading up to about 50% and a precise chip synchronization is required for QS-DBPSK. In a SRF channel, it is also shown that QS-DBPSK much outperforms MOS in a slow fading channel but MOS has a performance gain against the large Doppler shift.

As mobile communication systems have been widespread and the needs for new service grows, IMT-2000 systems have largely been researched and developed for standardization. Among them, Wideband CDMA (WCDMA) solution is standardized in the 3rd Generation Partnership Project. In WCDMA system, voice and high data rate services are supported through dedicated traffic channels, however other packet based services with short duration are provided by using common channels. Data users are allowed to transmit a short message based on a contention manner via the reverse common channels. The basic Common Packet Channel (CPCH) mechanism has been proposed and adopted for accessing common channels. It consists of three phases: Random Access Phase, Collision Resolution Phase, and Data Transmission Phase. To enhance the resource usage efficiency, the CPCH mechanisms with channel assignment or status monitoring have been proposed. They can improve the performance, however increase the system complexity. Up to now, performances of the CPCH mechanisms have been discussed and studied based on computer simulations. Numerical results have been obtained by using S-G analysis, but they are different from simulation results. In this paper, we analyze the CPCH mechanisms by using the Equilibrium Point Analysis (EPA). And we compare computer simulation results with analytical results.

Fuzzy inference abilities were implemented to electromagnetic problems for the first time by the authors. After very successful results of applying the developed fuzzy modeling method to input impedance of a general monopole antenna, in this paper classifying the engineering electromagnetic problems simply, we apply the abilities of the proposed fuzzy inference method to make a qualitative model for transmission lines as a general example for a certain category of problems. The proposed approach starts from observing the problem through the window of human direct understandings and uses some parameters (as calculation base) evaluated basic for modeling process. It is shown that because of using this novel view point, a very simple fuzzy system based on new parameters may model the behavior of a transmission line in general form. The knowledge of each variable can be extracted and saved as simple curves individually, through continuing to make several models considering the desired variable as parameter. Finally, it is shown that the proposed method works even in highly nonuniform transmission line cases without changing in structure and complexity.

Multilayer feedforward neural network (MFNN) trained by the backpropagation (BP) algorithm is one of the most significant models in artificial neural networks. MFNNs have been used in many areas of signal and image processing due to high applicability. Although they have been implemented as analog, mixed analog-digital and fully digital VLSI circuits, it is still difficult to realize their hardware implementation with the BP learning function efficiently. This paper describes a special BP algorithm for the logic oriented neural network (LOGO-NN) which we have proposed as a sort of MFNN with quantized weights and multilevel threshold neurons. Both weights and neuron outputs are quantized to integer values in LOGO-NNs. Furthermore, the proposed BP algorithm can reduce high precise calculations. Therefore, it is expected that LOGO-NNs with BP learning can be more effectively implemented as digital type circuits than the common MFNNs with the classical BP. Finally, it is shown by simulations that the proposed BP algorithm for LOGO-NNs has good performance in terms of the convergence rate, convergence speed and generalization capability.

Equivalent-time sampling is a well-known technique to capture repetitive signals at finer time intervals than a sampling clock cycle time and it is widely used to implement waveform measurement with high time resolution. There are three techniques for implementing its time base (i.e., sequential sampling, random sampling and coherent sampling), and they have their respective advantages and disadvantages. In this paper we propose a new coherent sampling system which incorporates a pretrigger and time jitter reduction function for a fluctuating input signal which a random sampling system has, while maintaining the waveform recording efficiency of a conventional coherent sampling system. We also report on a technique for measuring a reference trigger time period accurately which is necessary to implement the proposed sampling system, and show its effectiveness through numerical calculations of its data recording time.