Quality of Service (QoS) provisioning is a new but challenging research area in the field of Mobile Ad hoc Network (MANET) to support multimedia data communication. However, the existing QoS routing protocols in ad hoc network did not consider a major aspect of wireless environment, i.e., mutual interference. Interference between nodes belonging to two or more routes within the proximity of one another causes Route Coupling. This can be avoided by using zone-disjoint routes. Two routes are said to be zone disjoint if data communication over one path does not interfere with the data communication along the other path. In this paper, we have proposed a scheme for supporting priority-based QoS in MANET by classifying the traffic flows in the network into different priority classes and giving different treatment to the flows belonging to different classes during routing so that the high priority flows will achieve best possible throughput. Our objective is to reduce the effect of coupling between routes used by high and low priority traffic by reserving zone of communication. The part of the network, used for high priority data communication, i.e, high priority zone, will be avoided by low priority data through the selection of a different route that is maximally zone-disjoint with respect to high priority zones and which consequently allows contention-free transmission of high priority traffic. The suggested protocol in our paper selects shortest path for high priority traffic and diverse routes for low priority traffic that will minimally interfere with high priority flows, thus reducing the effect of coupling between high and low priority routes. This adaptive, priority-based routing protocol is implemented on Qualnet Simulator using directional antenna to prove the effectiveness of our proposal. The use of directional antenna in our protocol largely reduces the probability of radio interference between communicating hosts compared to omni-directional antenna and improves the overall utilization of the wireless medium in the context of ad hoc wireless network through Space Division Multiple Access (SDMA).

Recent advances in measurement techniques for microwave active devices and circuits are reviewed in this paper. The R&D activities have been devoted aggressively how to characterize nonlinear performance of high power devices and circuits. They are pulsed I-V, a variety of load-pull measurements, probing, sampling, and sensing techniques, supported by the recent significant advances in DSP (Digital Signal Processing), RF components, semiconductor devices, etc. The recent advances in vector network analyzers are of our great interest. They are (a) multi-port vector network analyzers for characterizing mixers, differential devices, packaged components, electronic package characterization, and multi-layer transmission lines, and (b) EO (Electro-Optic) modulated vector network analyzers for characterizing electronic performance of EO devices with the aid of EO modulators and photonic probes. In addition, probing, sampling, and sensing techniques have made great progress to directly measure electromagnetic field, time-domain voltage waveform, and temperature in small spot areas. In this paper, some topics related to these measurement techniques are briefly reviewed. Then the existing and future issues for characterization and measurement techniques of microwave active devices and circuits are discussed.

A hybrid method-of-moments (MoM) and immittance approach for efficient and accurate analysis of printed slots and strips of arbitrary shape in layered waveguide for various applications has been proposed. An impedance-type MoM is formulated from the electric field integral equation (EFIE) for printed strip case and an admittance-type MoM is formulated from the magnetic field integral equation (MFIE) for the printed slot case, using the Galerkin's technique. Immittance approach has been used to calculate spectral dyadic Green's functions for the layered waveguide. For efficient analysis of large and complex structures, equivalent circuit parameters of a block are first extracted and complete structure is analyzed through cascaded ABCD matrices. The equivalent circuit characterization of printed strip and slot in layered waveguide has been done for the first time. Finite periodic structure loaded with printed strips has been investigated and it shows the electromagnetic bandgap (EBG) behavior. The electromagnetic (EM) program hence developed is checked for its numerical accuracy and efficiency with results generated with High-frequency structure simulator (HFSS) and shows good performance.

A novel method for measuring microwave reflection coefficients without the open and load standards is proposed. In this method, a single probe is inserted into an air line and the output wave is detected by a vector detector. Offset shorts are used for the calibration. The measurement system is constructed using 7 mm coaxial line and APC7 connectors. The result of the measurement in the frequency range 1-9 GHz shows the possibility of the proposed method. All the major systematic errors can be estimated from the data that is easily obtainable.

Recently, low-density parity-check (LDPC) codes have attracted much attention. LDPC codes can achieve the near Shannon limit performance like turbo codes. For the LDPC codes, the reduced complexity decoding algorithms referred to as uniformly most powerful (UMP) BP- and normalized BP-based algorithms were proposed for BPSK on an additive white Gaussian noise (AWGN) channel. The conventional BP and BP-based algorithms can be applied to BPSK modulation. For high bit-rate transmission, multilevel modulation is preferred. Thus, the BP algorithm for multilevel modulations is proposed in . In this paper, we propose the BP algorithm with reduced complexity for multilevel modulations, where the first likelihood of the proposed BP algorithm is modified to adjust multilevel modulations. We compare the error rate performance of the proposed algorithm with that of the conventional algorithm on AWGN and flat Rayleigh fading channels. We also propose the UMP BP- and normalized BP-based algorithms for multilevel modulations on AWGN and flat Rayleigh fading channels. We show that the error rate performance of the proposed BP algorithm is almost identical to that of the algorithm in, where the decoding complexity of the proposed BP algorithm is less than that of the algorithm in. We also show that the proposed BP-based algorithms can achieve the good trade-off between the complexity and the error rate performance.

Support vector algorithms try to maximize the shortest distance between sample points and discrimination hyperplane. This paper suggests the total margin algorithms which consider the distance between all data points and the separating hyperplane. The method extends and modifies the existing algorithms. Experimental studies show that the total margin algorithms provide good performance comparing with the existing support vector algorithms.

We presented a wall charge controlled ac PDP driving method which has advantages of less number of sub-fields and no dynamic false contours compared to the conventional driving method. In this method, a sub-field exhibited different light intensity according to the initial wall charge quantity set during the address period. Even though one can set 10 different wall charge states by changing the data pulse widths, we decided to define three states, 'on,' 'half-on,' and 'off.' By adding one state, the number of sub-field required to achieve 243 gray levels was reduced from 8 to 5. Furthermore, one can realize seven sub-fields, 255 gray level, complete stretched-out coding with which one can eliminate the dynamic false contours. Since this method can reduce number of sub-fields, it is suitable for higher resolution PDP's with more scan lines.

This paper considers the least-squares linear estimation problem of signals from randomly delayed observations when the additive white noise is correlated with the signal. The delay values are treated as unknown variables, modelled by a binary white noise with values zero or one; these values indicate that the measurements arrive in time or they are delayed by one sampling time. A recursive one-stage prediction and filtering algorithm is obtained by an innovation approach and do not use the state-space model of the signal. It is assumed that both, the autocovariance functions of the signal and the crosscovariance function between the signal and the observation noise are expressed in a semi-degenerate kernel form; using this information and the delay probabilities, the estimators are recursively obtained.

We study the user reneging behavior in a hybrid Multimedia-on-Demand (MoD) system, which provides both interactive and batch services. Reneging happens when users become reluctant to wait for a batch service and quit the service. They will either join the interactive service or quit the system. In this letter, the analytical model for the MoD system with user reneging behavior is provided. Numerical results show that the reneging behavior has a significant impact on system performance. Therefore, careful consideration on reneging behavior should be made for designing a MoD system.

In mobile multimedia environment, it is very important to minimize handoff latency due to mobility. In terms of reducing handoff latency, Hierarchical Mobile IPv6 (HMIPv6) can be an efficient approach, which uses a mobility agent called Mobility Anchor Point (MAP) in order to localize registration process. However, MAP can be a single point of failure or performance bottleneck. In order to provide mobile users with satisfactory quality of service and fault-tolerant service, it is required to cope with the failure of mobility agents. In, we proposed Robust Hierarchical Mobile IPv6 (RH-MIPv6), which is an enhanced HMIPv6 for fault-tolerant mobile services. In RH-MIPv6, an MN configures two regional CoA and registers them to two MAPs during binding update procedures. When a MAP fails, MNs serviced by the faulty MAP (i.e., primary MAP) can be served by a failure-free MAP (i.e., secondary MAP) by failure detection/recovery schemes in the case of the RH-MIPv6. In this paper, we investigate the comparative study of RH-MIPv6 and HMIPv6 under several performance factors such as MAP unavailability, MAP reliability, packet loss rate, and MAP blocking probability. To do this, we utilize a semi-Markov chain and a M/G/C/C queuing model. Numerical results indicate that RH-MIPv6 outperforms HMIPv6 for all performance factors, especially when failure rate is high.

Multiuser diversity, identified by recent information theoretic results, is a form of diversity inherent in a wireless network. The diversity gain is obtained from independent time-varying fading channels across different users. The main practical issue in multiuser diversity is lack of Quality of Service (QoS) guarantees. This study proposes a wireless scheduling algorithm named MUDSEQ for downlink channels exploiting multiuser diversity under explicit QoS constraints. The numerical results demonstrate that the novel algorithm can yield non-negligible diversity gain even under tight QoS constraints and little scattering or slow fading environments. Additionally, a system framework for dynamic resource allocation based on the proposed algorithm is developed.

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.

Recent investigations using magnetic resonance imaging (MRI) of human speech organs have opened up new avenues of research. Visualization of the speech production system provides abundant information on the physiological and acoustic realization of human speech. This article summarizes the current status of MRI applications with respect to speech research as well as our own experience of discovery and re-evaluation of acoustic events emanating from the vocal tract and physiological mechanisms.

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.

Future IP networks will provide multi-class-services that have multiple levels of Quality of Services (QoS) at different prices. One of the issues for the network service provider (NSP) will be how to profit by providing them. This paper proposes a scheme that maximizes the profit of the NSP by controlling the service-list under the constraint of the available network resources. We introduce a model in which the users' selection from among the multiple classes is influenced not just by the price and QoS of one class, but the prices and QoS levels of all classes. In short, the user's selection involves a balance between the price and QoS levels of all classes. To model the users' class choice, we adopt discrete choice analysis; it can estimate the model parameters such that the model fits actual choice data. This paper proposes a functional framework that consists of User Choice Model Function, Original Demand Forecast Function, and Service-list Determination Function. The proposed model has the advantage of following actual changes adaptively. Effectiveness of the proposed scheme is evaluated by computer simulation for a multiple class service; even if the real parameters are changed, the proposal can follow the change and provide the optimal service-list that maximizes profit adaptively.

Currently, the IP infrastructure is widely available, which reduces the deployment and implementation cost of IP-based wireless networks. But, the corresponding researches on Quality of Service (QoS) in wireless mobile networks have not been done sufficiently. Although All-IP network has gotten the spotlight for both wired and wireless integrated networks, it has some unavoidable defects as it is based on best-effort IP transmission technologies. By focusing our attention on this point, we propose an Advanced QoS Provisioning Scheme (AQPS) that can provide satisfied QoS for multimedia applications based on accurate traffic measurements. With the AQPS, we consider 1xEV-DV system as a target system, extract QoS requirements for this system, and develop a corresponding QoS provisioning method using CDMA traffic measurement. The AQPS is designed to estimate delay parameters between a mobile station and a BTS (or repeater), verify whether the mobile station can support RSVP or not, operate a corresponding scheduler, and assign appropriate time slots based on traffic classes. One of the interesting results of this research is that the capability of distinguishing BTS' traffic from repeater's traffic helps save implementation cost of CDMA systems. The AQPS has been verified to outperform the existing system used in the current CDMA network and to efficiently manage network resources by computer simulations.

Service differentiation has been a subject of research for the past few years in the IETF; and in the current Internet, IP flows are mostly treated in a best-effort approach. However, for next generation networks it is expected that users would like to obtain service differentiation based on their preferences or profiles as well as the different types of multimedia they opt to receive or send. In addition, current Quality of Service (QoS) provisioning architectures have been designed mostly for the fixed networks without taking into consideration the wireless or radio links special requirements, such as low bandwidth availability, error prone communications, etc. In this paper we propose a QoS provisioning architecture for next generation networks that uses a hybrid approach to deal with both the wireless and wired (fixed) part of the network. For administering the scarce resource of the radio environment, we have developed a resource allocation algorithm based on micro-economic principles that uses associated piecewise linear utility functions which describe the benefit a user receives from the allocation of various amounts of resource. For the wired part of the network we have also developed a Core-Stateless Utility based Rate allocation Framework (SURF) for performing traffic policing where the flow's requirements are expressed using utility functions. The core routers maintain no per-flow state and implement a simple packet level admission control algorithm that is based on a threshold utility value that is computed dynamically. To tie in these two mechanisms, we developed a signaling mechanism that collect network statistics when a user starts a call and a QoS administrator entity (or Broker) perform the computations for allocating resources based on the information of available resources in the fixed and the wireless sections of the network. A comparison between the hybrid approach and the SURF approach to show the performance of the proposed architecture is presented later in the paper.

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.

Finite-ground microstrip line (FGMSL) open-end discontinuities are characterized via a self-calibrated method of moments (MoM) as a unified circuit model with a fringing capacitance and radiation conductance. By integrating the short-open calibration (SOC) procedure into a determinant MoM, the model parameters are extracted without needing the alternative port impedance. Regardless of non-ideal voltage sources, extracted parameters are observed to achieve a stable convergence as the feeding line is sufficiently extended. After extracted capacitance of a FGMSL open-end with equal strip and finite-ground widths are validated against its traditional MSL counterpart with infinite ground, extensive results are given to originally demonstrate that the capacitance increases as a decelerated function of the finite-ground width and length while the conductance is negligibly small as compared with its imaginary part.

This paper proposes a noise robust speech recognition method using prosodic information. In Japanese, the fundamental frequency (F0) contour represents phrase intonation and word accent information. Consequently, it conveys information about prosodic phrases and word boundaries. This paper first describes a noise robust F0 extraction method using the Hough transform, which achieves high extraction rates under various noise environments. Then it proposes a robust speech recognition method using multi-stream HMMs which model both segmental spectral and F0 contour information. Speaker-independent experiments are conducted using connected digits uttered by 11 male speakers in various kinds of noise and SNR conditions. The recognition error rate is reduced in all noise conditions, and the best absolute improvement of digit accuracy is about 4.5%. This improvement is achieved by robust digit boundary detection using the prosodic information.