In this paper, under the assumption that all the eigenvalues of a linear nominal discrete time-delay system lie within a specified ring region, a sufficient condition is proposed to preserve the assumed property when the structured parameter perturbations are added into the linear nominal discrete time-delay system. For the case of eigenvalue-clustering in a circular region, and for the case of not including time delays, the presented sufficient condition is mathematically proved to be less conservative than those reported recently in the literature.

This paper proposes a synchronous acquisition method using correlation mapping by multiple-dwell detection considering false alarm penalty in M-ary/SS systems. In the method, first, the correlation value between a received signal and each assigned sequence in an M-ary/SS system in some short duration is calculated for each phase and stored in the mapping. Second, the maximum correlation value of each phase in the mapping is selected and arranged, then, the first probable synchronous phase is decided by the maximum one in these values. Simultaneously, data demodulation starts. Next, to recognize the synchronous phase, i.e., as considering false alarm penalty, the correlation values are calculated in longer duration, and the second probable phase with high reliability can be obtained by suppression of noise to signal level. Finally, if the second synchronous phase is different from the first one, the second one is reset. By this method, a short acquisition time and high reliability of acquisition can be achieved. The improvement of acquisition time and the optimal combination values of dwelling time, which is duration to calculate the correlation, are shown for several conditions in asynchronous M-ary/SSMA.

We describe a simplified receiver structure having several receiving antennas (i.e., an adaptive array antenna system) and using time-division-multiplexing (TDM) signal processing. Three simplified receiver structures were investigated for use in the antenna system. To confirm the feasibility of using a TDM receiver, both a TDM receiver and a conventional adaptive array receiver were constructed for testing. In our proposed system, several repetitions of the constant modulus algorithm (CMA) are used to reduce co-channel interference (CCI). The frame format used for both receivers was the same as that of the personal handy phone system in Japan. The laboratory testing was done using a fading simulator to enable measurement of the bit error rate. The results are very promising and show the feasibility of the TDM receiver.

In distributed network systems, it is one of the most important problems how to assign the files to servers in view of cost and delay. It is obvious that there is a trading-off relationship between costs and delays in these systems. In order to evaluate the optimization that the total cost is minimized subject to the total delay, we have presented the Optimal File Allocation Model as 0-1 integer programming, and have investigated the general characteristics in distributed systems. In this model, we have introduced many cost and delay parameters to evaluate the total cost and delay in the system more exactly. In constructing practical systems, it is necessary to investigate the weight and the contribution of each parameter to the total cost. It is very useful to show how to estimate cost and delay parameters on the basis of this analysis. In this paper, we analyze the sensitivity of these parameters and make clear the influence between principal parameters.

The inter-cell interference between uplinks and downlinks in CDMA packet communication systems, employing a shared-TDD scheme, is evaluated under cellular environments. It is found that interference between base stations rarely degrades uplink throughput, but interference between mobile stations substantially degrades downlink throughput. A transmission power control scheme is proposed to improve the downlink throughput. The proposed scheme increases the transmission power of downlink packets when they are re-transmitted, and thus, improves the signal-to-interference ratio of the downlink re-transmission packets. Computer simulation shows that this scheme increases downlink throughput without sacrificing the uplink throughput until the uplink throughput reaches a maximum value.

Among several multiprocessor topologies, two-dimensional (2D) mesh topology has become popular due to its simplicity and efficiency. Even though a number of scheduling and processor allocation schemes for 2D meshes have been proposed in the literature, little study has been done aimed for real-time environment. In this paper, we propose an on-line scheduling and allocation scheme for real-time tasks that require the exclusive use of submeshes in 2D mesh system. By effectively manipulating the information on allocated or reserved submeshes, the proposed scheme can quickly identify the earliest available time of a free submesh for a newly arrived task. We employ a limited preemption approach to reduce the complexity of the search for a feasible schedule. Computer simulation reveals that the proposed scheme allows high throughput by decreasing the number of tasks rejected.

Recent progress in ultrahigh bit rate optical transmission using time-division multiplexing and wavelength-division multiplexing is described. Latest over 1 Tbit/s transmission experiments are shown and the possibility of higher bit rate transmission is discussed.

Recent progress in ultrahigh bit rate optical transmission using time-division multiplexing and wavelength-division multiplexing is described. Latest over 1 Tbit/s transmission experiments are shown and the possibility of higher bit rate transmission is discussed.

This paper proposes an area/time optimizing algorithm in a high-level synthesis system for control-based hardwares. Given a call graph whose node corresponds to a control flow of an application program, the algorithm generates a set of state-transition graphs which represents the input call graph under area and timing constraint. In the algorithm, first state-transition graphs which satisfy only timing constraint are generated and second they are transformed so that they can satisfy area constraint. Since the algorithm is directly applied to control-flow graphs, it can deal with control flows such as bit-wise processes and conditional branches. Further, the algorithm synthesizes more than one hardware architecture candidates from a single call graph for an application program. Designers of an application program can select several good hardware architectures among candidates depending on multiple design criteria. Experimental results for several control-based hardwares demonstrate effectiveness and efficiency of the algorithm.

This paper proposes a bandwidth allocation scheme which improves degradation of communication quality due to handoffs in mobile multimedia networks. In general, a multimedia call consists of several component calls. For example, a video phone call consists of a voice call and a video call. In realistic environments, each component call included in one multimedia call may have different requirements for quality-of-service (QoS) from each other, and priorities among these component calls often exist with respect to importance for communications. When the available bandwidth is not enough for a handoff call, the proposed scheme eliminates a low priority component call and defers bandwidth allocation for a component call whose delay related QoS is not strict. Moreover, in the allocation, the scheme gives priority to new calls and handoff calls over a deferred call and also performs bandwidth reallocation to eliminated component calls. By computer simulation, we evaluate the performance such as call dropping probability and show effectiveness of the proposed scheme.

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.

A new speech enhancement technique is proposed assuming that a speech signal is represented in terms of a linear probabilistic process and that a noise signal is represented in terms of a stationary random process. Since the target signal, i.e., speech, cannot be represented by a stationary random process, a Wiener filter does not yield an optimum solution to this problem regarding the minimum mean variance. Instead, a Kalman filter may provide a suitable solution in this case. In the Kalman filter, a signal is represented as a sequence of varying state vectors, and the transition is dominated by transition matrices. Our proposal is to construct the state vectors as well as the transition matrices based on time-frequency pattern of signals calculated by a wavelet transformation (WT). Computer simulations verify that the proposed technique has a high potential to suppress noise signals.

In this paper, we consider a mobile, multimedia, multihop (M3) ad hoc network. Key characteristics of this system are the mobility of users, energy constraints, and the need to operate without a fixed (wired or wireless) infrastructure. In this environment, with the advent of multimedia communications, the use of the cluster architecture has been revisited to support the resource reservation and Quality-of-Service routing. We proposed an access-based clustering protocol (ABCP) that allows the network to self-organize into a cluster architecture. Three advantages are claimed by ABCP. First, by the access-based criterion, it minimizes the overhead on cluster formation so that the protocol has short execution time and good scalability. Second, ABCP unifies the algorithms for cluster initialization and maintenance, i.e., the same set of clustering functions are used by a node regardless of whether it just becomes active or is in leaving its current cluster. Third, simulation results demonstrate that the cluster structure behaves stably amid topology changes compared with techniques previously proposed. Together with the access-based criterion, a multiple access scheme is also proposed for the broadcast of control messages.

Broadcast data delivery is attractive for large-size data distribution where a large user community is connected to a server through a network. It is important to consider a broadcast scheduling method which minimizes the average response time. The scheme should also guarantee the expected waiting time at the time of request. In this paper, we propose a method which divides all titles into several groups and assigns FIFO to each group. The proposed method can guarantee the waiting time for each user at his request, and is superior to FIFO (in high load) and a fixed allocation method (in low load).

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

Due to the lack of support in mobility and quality of service, today's IP-based networks have some inherent limitations for delivering multimedia services in a mobile environment. In the past few years, these issues have been addressed in the research community and the resulting techniques are being standardised. However, these developments have been done in isolation and become incompatible with each other. In addition to these technical issues, the future infrastructure for charging and accounting mobile multimedia services is expected to be increasingly complicated. In this paper we present a home-proxy based framework which can facilitate the integration of mobility support and QoS management. Furthermore, it enables centralised accounting, which simplifies the cost recovery processes of roaming services. To prove the viability of our design, we have built and tested this framework on a DiffServ wide-area backbone using an MP3 streaming application.

This paper proposes a very small on-chip multimedia real-time OS for embedded system LSIs, and demonstrates its usefulness on MPEG-2 multimedia applications. The real-time OS, which has a conditional cyclic task with suspend and resume for interacting hardware (HW) / software (SW) of embedded system LSIs, implements the minimum set of task, interrupt, and semaphore managements on the basis of an analysis of embedded software requirements. It requires only about 2.5 Kbytes memory on run-time, reduces redundant conventional cyclic task execution steps to about 1/2 for HW/SW interactions, and provides sufficient performance in real-time through implementing two typical embedded softwares for practical multimedia system LSIs: an MPEG-2 system protocol LSI and an MPEG-2 video encoder LSI. This on-chip multimedia real-time OS with 2.5 Kbyte memory will be acceptable for future multimedia embedded system LSIs.

Currently, the issues in Quality of Service, fairness and pricing strategies should have expedited the emergence of service differentiation in wireless access networks. In this paper, we propose a novel scheduling algorithm, called the Wireless Differentiated Fair Queueing (WDFQ) algorithm, to accommodate such need by providing delay/jitter controls, and fair residual bandwidth sharing for real-time and non-real-time traffic streams simultaneously. We show that the WDFQ scheme can achieve excellent performance, including timely delivery of real-time traffic, virtually error-free transmission of non-real-time traffic, and fair usage of channel bandwidth among remote stations. In addition, the location-dependent channel error property, as appeared in most wireless networks, are considered in the model and the temporary short error bursts are compensated by credits of bandwidth. The simulation results suggest that the length of retransmission period should be adapted to the error length to achieve good performance and maintain low implementation complexity.

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.

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.