FIR digital filters composed of parallel multiple subfilters are proposed. A binary expression of an input signal is decomposed into multiple shorter words, which drive the subfilters having different length. The output error is evaluated by mean squared and maximum spectra. A fast algorithm is also proposed to determine optimal filter lengths and coefficients of subfilters. Many examples confirm that the proposed filters generate smaller output errors than conventional filters under the condition of specified number of multiplications and additions in filter operations. Further, multiplier and adder structures (MAS) to perform the operations of the proposed filters are also presented. The number of gates used in the proposed MAS and its critical path are estimated. The effectiveness of the proposed MAS is confirmed.

This paper introduces a new analytic method that uses modified state equations to evaluate the performance of PCSD (Packet Channel Sharing protocol for DCA systems) with the goal of increasing the spectrum efficiency of DCA systems by realizing channel sharing between circuit-switched calls and packets. The results of this analysis show that PCSD is more suitable for microcellular systems than cellular systems, and that PCSD system performance improves as the average holding time of circuit-switched calls increases. Moreover, this paper proposes a novel scheme to determine the optimum release delay time of packet channels in order to achieve high throughput for packets as well as high channel capacity for circuit-switched calls. The proposed scheme shows that the optimum release delay time for PHS (Personal Handy-phone System) is greater than 60 frames and less than 100 frames.

This paper studies a congestion control scheme in integrated variable bit-rate video, audio and data (e. g. , image or text) communications, where each video stream is synchronized with the corresponding audio stream. When the audio and video streams are output, media synchronization control is performed. To avoid congestion, we employ a dynamic video resolution control scheme which dynamically changes the video encoding rate according to the network loads. By simulation, the paper evaluates the performance of the scheme in terms of throughput, loss rate, average delay, and mean square error of synchronization. Numerical results show the effectiveness of the scheme.

A more judicious choice of trial functions to implement the Improved Circuit Theory (ICT) application to multi-element antennas is achieved. These new trial functions, based on Tai's modified variational implementation for single element antennas, leads to an ICT implementation applicable to much longer co-planar dipole arrays. The accuracy of the generalized impedance formulas is in good agreement with the method of moments. Moreover, all these generalized formulas including the radiation pattern expressions are all in closed-form. This leads to an ICT implementation which still requires much shorter CPU time and lesser computer storage compared to method of moments. Thus, for co-planar dipole arrays, the proposed implementation presents a relatively very efficient method and would therefore be found useful in applications such as CAD/CAE systems.

In the multipath mobile channel, the received signal suffers from both the fluctuation in the received field intensity caused by fading and waveform distortion caused by the echo. Diversity reception using multiple spaced antennas is an effective method to compensate for fading, while echo cancellation with an adaptive array is good at compensating for waveform distortion. In this paper, an adaptive switching echo cancellation/diversity reception method to compensate for both waveform distortion and fading is proposed. The proposed switching reception monitors the impacts of channel conditions on received signal and then one of an echo canceller and a diversity receiver is selected accordingly to compensate the channel. The compensation performance of the proposed switching reception in terms of both average DUR (Desired to Undesired signal Ratio) and the probability of DUR below a threshold value is investigated with computer simulation. The results show that the adaptive switching echo cancellation/diversity reception has realized the advantages of both adaptive echo cancellation and diversity reception.

In this paper, we develop a novel method for tuning parameters known as the sensitivity parameters of membership functions used in a fuzzy classifier. The proposed method performs tuning by solving a set of inequalities. Each inequality represents a range of the ratio of the sensitivity parameters between the corresponding pair of classes. The range ensures the maximum classification rate for data of the two corresponding classes used for tuning. First, we discuss how such a set of inequalities is derived. We then propose an algorithm to solve the derived set of inequalities. We demonstrate the effectiveness of the proposed tuning method using two classification problems, namely, classification of commonly used iris data, and recognition of vehicle licence plates. The results are compared with those obtained by using the existing tuning method and with those by neural networks.

The paper is focused on the architectural and technological solutions that will allow the transition from small to huge capacity ATM Switching Systems. This path starts from the industrial nodes available today and will arrive at the photonic switching architecture. The progressive introduction of photonics has already started with the use of optical interconnections in ATM nodes of hundreds of Gbit/s. A balanced use of microelectronics and photonics is the correct answer to the Terabit/s switching system challenge. After presenting a modular ATM Switching System, some technological solutions like Multichip Modules and Optical Interconnections are presented in order to explain how node capacity can be expanded. Some results of the research activity on photonic Switching are finally shown in order to exploit the great attitude of this technique to obtain very high throughput nodes.

Detection of facial emotions are mainly addressed by computer vision researchers based on facial display. Also detection of vocal expressions of emotions is found in research work done by acoustic researchers. Most of these research paradigms are devoted purely to visual or purely to auditory human emotion detection. However we found that it is very interesting to consider both of these auditory and visual informations together, for processing, since we hope this kind of multimodal information processing will become a datum of information processing in future multimedia era. By several intensive subjective evaluation studies we found that human beings recognize Anger, happiness, Surprise and Dislike by their visual appearance, compared to voice only detection. When the audio track of each emotion clip is dubbed with a different type of auditory emotional expression, still Anger, Happiness and Surprise were video dominant. However Dislike emotion gave mixed responses to different speakers. In both studies we found that Sadness and Fear emotions were audio dominant. As a conclusion to the paper we propose a method of facial emotion detection by using a hybrid approach, which uses multimodal informations for facial emotion recognition.

This paper proposes a time-interval data model in which all temporal representation and operations can be expressed with time intervals. The model expresses not only real time intervals, in which an event exists, but also null time intervals, in which an event is suspended. We model the history of a real-world event as a composite time interval, which is defined in this paper. Operations on the composite time intervals are also defined, and it is shown how these operations can be used to express temporal constraints with time intervals.

Recently reported multidimensional geometrically uniform signal constellations (L MPSK and Decomposed-Lattice constellations) are joined in the term of Generalized Permutation Alphabets (GPA). Possibility of a binary isometric labeling of GPA's is completely characterized. An algorithm for constructing generating groups of PSK-type GPA is proposed. We show that this concept, when is extended to the lattice, gives rise to a class of new coset codes which perform out best codes listed in [11].

In addition to additive thermal noise, a received direct-sequence spread spectrum (DS/SS) signal may suffer from intersymbol interference (ISI) and interference caused by cochannel narrowband users or other narrowband radio frequency interference (RFI). This paper presents a workable solution for removing narrowband interference (NBI) and reducing ISI or inter-chip interference (ICI) when the communication channel can be modeled as an FIR filter and the NBI comes from multiple CW tones, an AR-modeled Gaussian process, or a BPSK signal. Unlike earlier solutions, the proposed scheme is capable of performing the functions of NBI-rejection, ISI/ICI suppression and data detection (code despreading) simultaneously. It is easy to implement and, more importantly, it yields lower bit error rate (BER) and smaller mean squared error (MSE).

We propose a spread-spectrum power line communication system considering the cyclic features of the noise in the lines. For this purpose, we model the noise as the sum of a time-invariant stationary process and two cyclostationary proceses, i.e., cyclic continuous noise and cyclic impulsive noise. The proposed system employs two different countermeasures to each of these two classes of cyclic noise. For the cyclic continuous noise, it uses multiple-processing-gain spread spectrum technique: the smaller processing gains are assigned for the periods with lower instantaneous noise power and the larger ones for the periods of higher noise power. Considering the cyclic impulsive noise, convolutional coding with interleaving is applied. In order to analyze the performance improvement due to the employment of multiple processing gains, we introduce a simple model of the continuous noise. The overall performance is evaluated by computer simulation with the actual noise wave-form measured in power lines.

A slotted ALOHA direct sequence spread spectrum system with random signatures is considered. The system is applicable in cases where a large number of terminals transmit to a single hub station like in cellular digital radio, personal mobile systems and wireless LANs. It is shown that significant improvements in packet throughput capacity are obtained if the adaptive receiver structures are used. Systems for the comparison are the spread spectrum slotted ALOHA system and the conventional slotted ALOHA system.

The H-polarized diffraction by a wedge consisting of perfect conductor and lossless dielectric is investigated by employing the dual integral equations. Its physical optics diffraction coefficients are expressed in a finite series of cotangent functions weighted by the Fresnel reflection coefficients. A correction rule is extracted from the difference between the diffraction coefficients of the physical optics field and those of the exact solution to a perfectly conducting wedge. The angular period of the cotangent functions is changed to satisfy the edge condition at the tip of the wedge, and the poles of the cotangent functions are relocated to cancel out the incident field in the artificially complementary region. Numerical results assure that the presented correction is highly effective for reducing the error posed in the physical optics solution.

Error locating codes were first presented in 1963 by J.K. Wolf and B.Elspas. Since then several code design methods have been proposed. However, their algebraic structure has not yet been clarified. It is apparent that necessary and sufficient conditions for error correcting/detecting codes can be expressed by Hamming distance, but, on the other hand, those for error locating codes cannot always be expressed only by Hamming distance. This paper presents necessary and sufficient conditions for error locating codes by using a newly defined metric and a function. The function represents the number of bytes where Hamming distance between corresponding bytes of two codewords has a certain integer range. These conditions show that an error locating code having special code parameters is an error correcting/detecting code. This concludes that error locating codes include existing bit/byte error correcting/detecting codes in their special cases.

In this paper an algorithm for numerical investigation of the transmission line having a generalized polygonal cross-section and open interface is proposed. Solution of the eigenmode problem is based on the method called the domain product technique, which employs a Mathieu function expansion and provides an efficient technique to the analysis of the structures with multiangular boundaries. An agreement at the obtained numerical results with existing data confirms the applicability of the theoretical analysis given in the paper.

In this Letter, we investigate the loss performance of a discrete-time single-server queueing system with periodic vacations, with which we are often confronted in traffic control, such as cell scheduling or priority control schemes, at ATM nodes. Explicit expressions are derived for the cell loss ratio in terms of the distribution of the buffer contents in an infinite capacity queue.

We investigate an importance sampling (IS) simulation of MMPP/D/1 queueing to obtain an estimate for the survivor function P(Q > q) of the queue length Q in the steady state. In Ref.[11], we studied the IS simulation of 2-state MMPP/D/1 queueing and obtained the optimal simulation distribution, but the mathematical fundation of the theory was not enough. In this paper, we construct a discrete time Markov chain model of the n-state MMPP/D/1 queueing and extend the results of Ref.[11] to the n-state MMPP/D/1. Based on the Markov chain model, we determine the optimal IS simulation distribution fo the n-state MMPP/D/1 queueing by applying the large deviations theory, especially, the sample path large deviations theory. Then, we carry out IS simulation with the obtained optimal simulation distribution. Finally, we compare the simulation results of the IS simulation with the ordinary Monte Carlo (MC) simulation. We show that, in a typical case, the ratio of the computation time of the IS simulation to that of the MC simulation is about 10-7, and the 95% confidence interval of the IS is slightly improved compared with the MC.

We present theoretical results on the convergence of iterative methods for the solution of linear differential-algebraic equations arising form circuit simulation. The iterative methods considered include the continuous-time and discretetime waveform relaxation methods and the Krylov subspace methods in function space. The waveform generalized minimal residual method for solving linear differential-algebraic equations in function space is developed, which is one of the waveform Krylov subspace methods. Some new criteria for convergence of these iterative methods are derived. Examples are given to verify the convergence conditions.

Distributed multimedia applications are often sensitive to the Quality of Service (QoS) provided by the communication network. They usually require guaranteed QoS service, so that real-time communication is possible. However, searching a route with multiple QoS constraints is known to be a NP-complete problem. In this paper, we propose a new simple and efficient distributed QoS routing algorithm, called "DQoSR," for supporting real-time communication in high-speed networks. It searches a route that could guarantee bandwidth, delay, and delay jitter requirements. Routing decision is based only on the modified cost, hop and delay vectors stored in the routing table at each node and its directly connected neighbors. Moreover, DQoSR is proved to construct loop-free routes. Its worst case message complexity is O(|V|2), where |V| is the number of nodes in the network. Thus DQoSR is fast and scales well to large networks. Finally, extensive simulations show that average rate of establishing successful connection of DQoSR is very near to optimum (the difference is less than 0.4%).