This paper deals with a simple and practical power loss analysis simulator, which can actually estimate the total power losses of three phase voltage-fed Auxiliary resonant commutation pole snubber assisted soft switching inverter as well as hard-switching inverter. In order to estimate the switching power losses and conduction power losses of switching semiconductor power devices (IGBTs), which are incorporated into the inverters, the proposed practical simulator is making use of feasible switching power loss data tables and conduction power loss data tables, which are accumulated from the measured voltage and current operating waveforms of power semiconductor switching devices. The practical effectiveness of feasible simulation technique and power loss evaluations for power electronic conversion circuits and systems are confirmed by the simulation and experimental results basis under the conditions of soft switching and hard switching sinusoidal PWM schemes.

Classical models of speech recognition assume that a detailed, short-term analysis of the acoustic signal is essential for accurately decoding the speech signal and that this decoding process is rooted in the phonetic segment. This paper presents an alternative view, one in which the time scales required to accurately describe and model spoken language are both shorter and longer than the phonetic segment, and are inherently wedded to the syllable. The syllable reflects a singular property of the acoustic signal -- the modulation spectrum -- which provides a principled, quantitative framework to describe the process by which the listener proceeds from sound to meaning. The ability to understand spoken language (i.e., intelligibility) vitally depends on the integrity of the modulation spectrum within the core range of the syllable (3-10 Hz) and reflects the variation in syllable emphasis associated with the concept of prosodic prominence ("accent"). A model of spoken language is described in which the prosodic properties of the speech signal are embedded in the temporal dynamics associated with the syllable, a unit serving as the organizational interface among the various tiers of linguistic representation.

A CMOS DFE (decision feedback equalization) receiver with a clock-data skew compensation was implemented for the SSTL (stub-series terminated logic) SDRAM interface. The receiver consists of a 2 way interleaving DFE input buffer for ISI reduction and a X2 over-sampling phase detector for finding the optimum sampling clock position. The measurement results at 1.2 Gbps operation showed the increase of voltage margin by about 20% and the decrease of time jitter in the recovered sampling clock by about 40% by equalization in an SSTL channel with 2 pF 4 stub load. Active chip area and power consumption are 3001000 µm2 and 142 mW, respectively, with a 2.5 V, 0.25 µm CMOS process.

In this paper, a simple blind algorithm for a beamforming antenna is proposed. This algorithm exploits the property of cyclostationary signals whose cyclic autocorrelation function depends on delay as well as frequency. The cost function is the mean square error between the delay product of the beamformer output and a complex exponential. Exploiting the delay greatly reduces the possibility of capturing undesired signals. Through analysis of the minima of the non-quadratic cost function, conditions to extract a single signal are derived. Application of this algorithm to code-division multiple-access systems is considered, and it is shown through simulation that the desired signal can be extracted by appropriately choosing the delay as well as the frequency.

The rapid spread of cellular phones in recent years has facilitated not only voice communication but also Internet access via the cellular phone system, and in addition, subscriber demand has led to a diversification in the services provided. One service in high demand is the seamless use of cellular phones in both public and private wireless network areas. In the data world, there is already such an application in the form of public and private use of wireless LAN. However, an increase in the number of users would require the realization of low-cost, easy-to-install very small base stations (VSBS) that use the frequency band efficiently in order to allow private use of ordinary cellular phones. To bring such VSBS into effect, a technology that autonomously selects frequencies which do not interfere with the public communication system from out of the publicly used frequency band is essential for turning such VSBS into reality. This paper proposes a frequency selection algorithm that actively uses cellular phone features such as frequency selection and received signal level measurement, and discusses the results of verification experiments.

In communication systems such as mobile telecommunication systems and the Internet, resource sharing among coexisting real-time and non-real-time services is extremely important to provide multimedia services. This paper analytically investigates the performance of the packet data control algorithm proposed in. This algorithm efficiently uses radio resources by utilizing the remaining capacity that is not used by real-time services. The state probability vectors and transition probability matrices of both the real-time and non-real-time services are first derived and then the delay characteristics, the outage probability of voice users, and the outage probability of data users are evaluated. A performance analysis with high bit-rate non-real-time services is also presented.

W-CDMA (Wideband-CDMA) is expected to play a significant role in the radio access technology of third-generation mobile telecommunication systems. In second-generation systems, voice traffic from each user has been transmitted mainly via the dedicated transport (radio) channel. In addition, the third-generation systems will efficiently accommodate data traffic based on packet transmission in the shared common transport channel. Therefore, data traffic can be transmitted via one of two types of data channels: i.e., dedicated channels or common channels. However, the channel selecting/switching scheme has not been standardized; thus, system architectures and algorithms of channel-switching schemes in the RNC (Radio Network Controller) are dependent on its vendors, and network operators must determine the parameter settings related to channel selection. In this paper, we will deal with aspects of the architecture in detail, and propose possible algorithms for channel selecting/switching for fundamental reference systems which meet the specifications of the RNC. We will then evaluate our algorithms by means of simulations, and discuss the impact of parameter settings on performance, in terms of packet loss probability and utilization of dedicated channels.

In a telecommunication network system, a scheme for reforwarding call-terminating setup messages (SETUP messages) is used to guard against their loss. We have developed a method for evaluating the loss probability of these reforwarding schemes. We started with a stochastic model in which the messages are reforwarded after a constant time span from the time that the first messages have been forwarded. This model corresponds to the finite-capacity BPP/M/1/m model. We showed a method for calculating the "timeout" probability. We then added an approximate method for calculating the loss probability. Finally, using the proposed methods, we clarified the existence of the best reforwarding timelag.

A novel resonator structure for the cut-off circular waveguide method is proposed to suppress the unwanted TE modes in the axial direction and TM modes in the radial direction. In this method, a dielectric plate sample is placed between two copper circular cylinders and clamped by two clips. The cylinder regions constitute the TE0m mode cut-off waveguides. The measurement principle is based on a rigorous analysis by the Ritz-Galerkin method. Many resonance modes observed in the measurement can be identified effectively by mode charts. In order to verify the validity of the novel structure for this method, the temperature dependences for three low-loss organic material plates were measured in the frequency range 40 to 50 GHz. It is found that modified polyolefin plates have comparable electric characteristics and low price, compared with PTFE plates. Moreover, it is verified that the novel resonator structure is effective in improvement of accuracy and stability in measurement. The measurement precisions are estimated within 1 percent for εr and within 15 percent for tan δ.

We present a method for recognition of continuous Korean Sign Language (KSL). In the paper, we consider the segmentation problem of a continuous hand motion pattern in KSL. For this, we first extract sign sentences by removing linking gestures between sign sentences. We use a gesture tension model and fuzzy partitioning. Then, each sign sentence is disassembled into a set of elementary motions (EMs) according to its geometric pattern. The hidden Markov model is adopted to classify the segmented individual EMs.

This paper deals with distributed procedures, caused by negation as failure through a network, where general logic programs are distributed so that they communicate with each other in terms of negation as failure inquiries and responses, but not in terms of derivations of SLD resolutions. The common variables as channels in share for distributed programs are not treated, but negation as failure validated in the whole network is the object for communications of distributed programs. We can define the semantics for the distributed programs in a network. At the same time, we have distributed proof procedures for distributed programs, by means of negation as failure to be implemented through the network, where the soundness of the procedure is guaranteed by the defined semantics.

This paper presents a novel design of connected digit patterns to achieve high accuracy text-prompted speaker verification over a cellular phone network. To reduce the error rate, a phoneme-balanced connected digit pattern for enrollment, and digit-sequence-preserving connected digit patterns for verification (i.e. patterns preserving partial digit sequences of the enrollment pattern) are proposed. In addition to these, a decision procedure using multiple patterns has been designed to overcome the low quality of cellular phone speech. Experimental results on cellular phone speech showed the phoneme-balanced patterns for enrollment and digit-sequence-preserving patterns for verification reduced more than 50% of equal error rate compared to the conventional method using randomly-selected and randomly-reordered digit patterns. The decision procedure reduced 60% of the error rate. In addition, this paper shows that verification patterns depending on the pattern of a preceding utterance reduced 10% of the error rate. Overall, the error rate obtained by the proposed method was 1% for 99% of clients and 95% of impostors.

In this paper, we explore a method to the problem of spoken document categorization, which is the task of automatically assigning spoken documents into a set of predetermined categories. To categorize spoken documents, subword unit representations are used as an alternative to word units generated by either keyword spotting or large vocabulary continuous speech recognition (LVCSR). An advantage of using subword acoustic unit representations to spoken document categorization is that it does not require prior knowledge about the contents of the spoken documents and addresses the out of vocabulary (OOV) problem. Moreover, this method works in reliance on the sounds of speech rather than exact orthography. The use of subword units instead of words allows approximate matching on inaccurate transcriptions, makes "sounds-like" spoken document categorization possible. We also explore the performance of our method when the training set contains both perfect and errorful phonetic transcriptions, and hope the classifiers can learn from the confusion characteristics of recognizer and pronunciation variants of words to improve the robustness of whole system. Our experiments based on both artificial and real corrupted data sets show that the proposed method is more effective and robust than the word based method.

Nagauta (長唄) is one of the classical styles of Japanese singing. It has very original and unique prosodic patterns, where abrupt and sharp changes of F0 are often observed at mora (Japanese speech unit) transitions. This F0 change is sometimes found even within a single mora. In this paper, we propose a model to synthesize this unique F0 pattern by considering the abrupt and sharp changes as grace notes. Nagauta's original scores contain no strict descriptions of tones and durations. Therefore, the baseline melody realized in a performance depends on the singer and it is difficult to predict the baseline melody by looking only at the scores. In this paper, the baseline melody is explicitly given to a singer in the form of the standard notation and the singer is asked to sing the song in Nagauta style. By taking the standard score as input, the proposed model simulates the F0 pattern generated by the singer under this condition. Further, this paper shows an interesting phenomenon about power movements at the sharp F0 changes. Acoustic analysis of Nagauta singing samples reveals that the sharp increases of F0 and the sharp decreases of power are synchronized. Although no discussion on physiological mechanisms of this phenomenon is done in this paper, another model is proposed to generate the unique power patterns. Evaluation experiments are done with young Japanese listeners and their results indicate high validity of the two proposed models.

This paper proposes two alternative approaches that do not make use of a dictionary but instead utilizes different types of learned features to segment words in a language that has no explicit word boundary. Both methods utilize decision trees as knowledge representation acquired from a training corpus in the segmentation process. The first method, a language-dependent technique, applies a set of constructed features patterns based on character types to generate a set of heuristic segmentation rules. It separates a running text into a sequence of small chunks based on the given patterns, and constructs a decision tree for word segmentation. The second method extracts statistics of character sequences from a training corpus and uses them as features for the process of constructing a set of rules by decision tree induction. The latter needs no linguistic knowledge. By experiments on Thai language, both methods achieve relatively high accuracy but the latter performs much better.

A novel thresholding algorithm for change detection in video sequences is proposed. The method is based on image differencing and the intensity distribution of a difference image. With a difference image between two consecutive images, we prepare a new image model for the distribution of stationary pixels. The distribution of moving pixels is then separated by extracting the distribution of stationary pixels from the overall distribution of the difference image. Pixels that exhibit a significant change in intensity are classified using a likelihood criterion. The proposed algorithm is tested on the standard MPEG sequences and verified to have reliable performance.

We propose Max-Plus Linear (MPL) systems with selective parameters that can describe a certain class of Timed Petri nets (TPN). In this class, selector and joint places are incorporated with Single-Input and Single-Output Timed Event Graph (SISO TEG) subnets. We confirm that the proposed controller effectively works taking into account practical constraints through a numerical example.

Band Division MC-CDM (BD-MC-CDM) has been proposed for high quality wireless communications and has been investigated in terms of link level performance. In this paper, we investigate frequency band and time slot selection technique from the viewpoint of system level performance in order to realize the efficient BD-MC-CDM system under cellular environments. Then a downlink frequency band and time slot selection scheme is proposed for cellular BD-MC-CDM systems. The proposed scheme selects transmission frequency band according to the signal-to-interference ratio (SIR) estimated by using the pilot signal at mobile stations and also selects transmission time slot by using the SIR threshold. Simulation results show that the proposed scheme improves the downlink throughput but degrades delay performance and it has a trade off between throughput and delay performance. By selecting suitable control parameters, the proposed scheme achieves the throughput improvement without sacrificing the delay performance.

This paper provides methods for construction of pairing-based cryptosystems based on non-supersingular elliptic curves.

The present state of IEC and JIS standards is reviewed on measurement methods of low-loss dielectric and high-tempera-ture superconductor (HTS) materials in the microwave and millimeter wave range. Four resonance methods are discussed actually, that is, a two-dielectric resonator method for dielectric rod measurements, a two-sapphire resonator method for HTS film measurements, a cavity resonator method for microwave measurements of dielectric plates and a cutoff circular waveguide method for millimeter wave measurements of dielectric plates. These methods realize the high accuracy sufficient for measurements of temperature dependence of material properties.