A new type of sound source segregation method using robot-mounted microphones, which are free from strict head related transfer function (HRTF) estimation, has been proposed and successfully applied to three simultaneous speech recognition systems. The proposed segregation method is executed with sound intensity differences that are due to the particular arrangement of the four directivity microphones and the existence of a robot head acting as a sound barrier. The proposed method consists of three-layered signal processing: two-line SAFIA (binary masking based on the narrow band sound intensity comparison), two-line spectral subtraction and their integration. We performed 20 K vocabulary continuous speech recognition test in the presence of three speakers' simultaneous talk, and achieved more than 70% word error reduction compared with the case without any segregation processing.

This paper deals with a universal coding problem for a certain kind of multiterminal source coding system that we call the complementary delivery coding system. In this system, messages from two correlated sources are jointly encoded, and each decoder has access to one of the two messages to enable it to reproduce the other message. Both fixed-to-fixed length and fixed-to-variable length lossless coding schemes are considered. Explicit constructions of universal codes and bounds of the error probabilities are clarified via type-theoretical and graph-theoretical analyses.

It is predicted that there will be a high demand for video applications in future wireless networks including wireless ad hoc networks. However, supporting video applications over mobile ad hoc networks is more complicated than with other networks due to the lack of support from a preinstalled infrastructure. In this paper, we tackle this problem by adopting the concept of multipoint-to-point video transmission used in wire-line networks. A novel framework designed with features to accommodate the characteristics of ad hoc networks is presented. There are two key features in our proposal. First, Multiple Description Coding (MDC) scheme is used for video coding to reduce the redundancy by avoiding the transmission of duplicate video frames. Second, the routing protocol is expanded to include finding disjoint routes from video sources to the receiver so that a single link breakage or a single intermediate node failure affects transmission from only the minimum number of nodes. Furthermore, the use of disjoint routes also enables the workload to be distributed more evenly within the network. A simulation study was carried out using NS-2 to demonstrate the performance of the proposed mechanism. We show that the proposed mechanism outperforms conventional point-to-point transmission, especially under conditions of high mobility.

A novel Adaptive Resource-based Probabilistic Search algorithm (ARPS) for P2P networks is proposed in this paper. ARPS introduces probabilistic forwarding for query messages according to the popularity of the resource being searched. A mechanism is introduced to estimate the popularity and adjust the forwarding probability accordingly such that a tradeoff between search performance and cost can be made. Using computer simulations, we compare the performance of ARPS with several other search algorithms. It is shown that ARPS performs well under various P2P scenarios. ARPS guarantees a success rate above a certain level under all circumstances, and enjoys high and popularity-invariant search success rate. Furthermore, ARPS adapts well to the variation of popularity, resulting in high efficiency and flexibility.

This paper presents a novel method to increase an impedance of a current source. The proposed circuit with a cascode and gain-boosting configuration is also presented. The operation has been confirmed by simulation using a 0.18 µm CMOS technology.

In real-time systems, deadline misses of the tasks cause a degradation in the quality of their results. To improve the quality, we have to allocate CPU utilization for each task adaptively. Recently, Buttazzo et al. address a feedback scheduling algorithm, which dynamically adjusts task periods based on the current workloads by applying a linear elastic task model. In their model, the utilization allocated to each task is treated as the length of a linear spring and its flexibility is described by a constant elastic coefficient. In this paper, we first consider a nonlinear elastic task model, where the elastic coefficient depends on the utilization allocated to the task. We propose a simple iterative method for calculating the desired allocated resource and derive a sufficient condition for the convergence of the method. Next, we apply the nonlinear elastic model to an adaptive fair sharing controller. Finally, we show the effectiveness of the proposed method by computer simulation.

In this paper, we propose an efficient resource discovery scheme for large-scale ubiquitous computing environments, which supports scalable semantic searches and load balancing among resource discovery resolvers. Here, the resources are described based on the concepts defined in the ontological hierarchy. To semantically search the resources in a scalable manner, we propose a semantic vector space and semantic resource discovery network in which the resources are organized based on their respective semantic distances. Most importantly, landmarks are introduced for the first time to reduce the dimensionality of the vector space. Computer simulation with CAN verifies the effectiveness of the proposed scheme.

A two-stage least square identification method is proposed for estimating ARMA (autoregressive moving average) coefficients from speech signals. A pulse-train like input sequence is often employed to account for the source effects in estimating vocal tract parameters of voiced speech. Due to glottal and radiation effects, the pulse train, however, does not represent the effective voice source. The authors have already proposed a simple but effective model of voice source for estimating AR (autoregressive) coefficients. This letter extends our approach to ARMA analysis to wider varieties of speech sounds including nasal vowels and consonants. Analysis results on both synthetic and natural nasal speech are presented to demonstrate the analysis ability of the method.

We introduce an adaptive subchannel, bit, and power allocation (ASBPA) algorithm to maximize the bandwidth efficiency of the mobile communication system that use orthogonal frequency division multiplexing (OFDM). We propose a suboptimal rate adaptive ASBPA algorithm that guarantees fairness in resource allocation and overcomes inherent co-channel interference (CCI) in the cellular system. Furthermore, we evaluate the maximum possible bandwidth efficiency of the cellular OFDM system achieved by the ASBPA algorithm which is practical to implement. Our simulation results show that the proposed algorithm outperforms the existing ones and achieves the cellular bandwidth efficiency of up to 5 b/s/Hz/cell. We also investigate some of the conditions that govern the bandwidth efficiency of the cellular OFDM system using the proposed ASBPA algorithm.

This paper proposes new resource management schemes for multiple data streams in an orthogonal frequency and space division multiplex access (OFSDMA) system using Radio-on-Fiber (RoF) ubiquitous antennas. The proposed schemes classify the services into some classes in which the number of sub-carriers is dynamically assigned according to the requested data size. The computer simulation results show that the proposed schemes improve the number of users satisfying the required bit error rate (BER) level as well as the average throughput and also show that the RoF ubiquitous antennas can improve system capacity.

This letter proposes a simple and efficient random-binning based distributed source coding (DSC) scheme for application to remote source estimation in wireless sensor networks. The scheme jointly encodes data from multiple sensors with side information. It achieves high coding efficiency and reduces power and bandwidth consumption.

This paper presents a scheme for accurately detecting the number of incident waves arriving at array antennas. The array antenna and MIMO techniques are developing as 4th generation mobile communication systems and wireless LAN technologies, and the accurate estimation of the propagation environment is becoming more important. This paper emphasizes the accurate detection of the number of incident waves; one of the important characteristics in multidirectional communication. There are some recent papers on accurate detection but they have problems of estimation accuracy or computational cost in severe environment like low SNR, small number of snapshots or waves with close angles. Hence, AIC and MDL methods based on statistics and information theory are still often used. In this paper, we propose an accurate estimation method of the number of arrival signals using the orthogonality of subspaces derived from preliminary estimation of signal subspace. The proposed method accurately estimates the number of signals also in severe environments where AIC and MDL methods can hardly estimate. We evaluate the performance of these methods through some computer simulation and experiments in anechoic chamber.

Viable techniques such as dynamic voltage scaling (DVS) provide a new design technique to balance system performance and energy saving. In this paper, we extend previous works on task assignment problems for a set of linear-pipeline tasks over a set of processors. Different from previous works, we revisit the problems with two additional system factors: deadline and energy-consumption, which are key factors in real-time and power-aware computation. We propose an O(nm2) time complexity algorithm to determine optimal task-assignment and speed-setting schemes leading to minimal energy consumption, for a given set of m real-time tasks running on n identical processors (with or without DVS supports). The same result can be extended to a restricted form of heterogeneous processor model. Meanwhile, we show that on homogeneous processor model more efficient algorithms can be applied and result in time complexity of O(m2) when m ≤ n. For completeness, we also discuss cases without contiguity constraints. We show under such cases the problems become at least as hard as NP-hard.

Recently, a word-valued source has been proposed as a new class of information source models. A word-valued source is regarded as a source with a probability distribution over a word set. Although a word-valued source is a nonstationary source in general, it has been proved that an entropy rate of the source exists and the Asymptotic Equipartition Property (AEP) holds when the word set of the source is prefix-free. However, when the word set is not prefix-free (non-prefix-free), only an upper bound on the entropy density rate for an i.i.d. word-valued source has been derived so far. In this paper, we newly derive a lower bound on the entropy density rate for an i.i.d. word-valued source with a finite non-prefix-free word set. Then some numerical examples are given in order to investigate the behavior of the bounds.

This paper proposes a network design scheme for Virtual Private Network (VPN) services. Traditionally, network design to compute required amount of resource is based on static point-to-point resource demand. This scheme is effective for traditional private line services. However, since VPN services allow multi-site connectivity for customers, it may not be appropriate to design a network based on static point-to-point resource demand. In particular, this scheme is not effective when the traffic pattern changes over time. Therefore, network design for VPN services introduces a new challenge in order to comply with traffic flexibility. There are conventional studies tackling this issue. In those studies, by defining a resource demand model considering flexibility, and designing the network based on this model, amount of resource required can be computed. However, there are some deficiencies in those studies. This paper proposes a new network design scheme, consisting of two components. The first one is a new resource demand model, created by extending conventional resource demand models, that can specify resource demand more precisely. The second one is a new network design algorithm for this resource demand model. Simulations are conducted to evaluate the performance of the proposed network design scheme, and the results show significant performance improvement against conventional schemes. In addition, deployment considerations of the proposed scheme are analyzed.

We consider coding for sources that output the symbols according to Poisson process from the viewpoint of real-time transmission. In order to reduce the transmission delay we avoid using input buffers. However, the lack of buffer causes overflow error. The theoretical relation between the transmission rate and the error probability is clarified. It is shown that the optimal code that minimizes the probability of error differs from the code that minimizes the expected codeword length. We also investigate the case of block coding as one of the applications of buffers.

This paper considers the universal coding problem for stationary ergodic sources with countably infinite alphabets. We propose modified versions of LZ77 and LZ78 codes for sources with countably infinite alphabets. Then, we show that for any source µ with Eµ[log X1]<∞, both codes are asymptotically optimum, i.e. the code length per input symbol approaches its entropy rate with probability one. Further, we show that we can modify LZ77 and LZ78 codes so that both are asymptotically optimal for a family of ergodic sources satisfying Kieffer's condition.

In this paper, we present a resource monitoring and selection method for rapid turnaround grid applications (for example, within 10 seconds). The novelty of our method is the distributed evaluation of resources for rapidly selecting the appropriate idle resources. We integrate our method with a widely used resource management system, namely the Monitoring and Discovery System 2 (MDS2), and compare our method with the original MDS2 in terms of the performance and the scalability. The performance is measured using a 64-node cluster of PCs and the scalability is analyzed using a theoretical model and the measured performance. The experimental results show that our method reduces the resource selection time by 82%, as compared with the original MDS2. The scalability analysis also indicates that our method can keep the resource selection time within 1 second, up to 500 nodes in local-area-network (LAN) environments. In addition, some simulation results are presented to estimate the impact of our method for wide-area-network (WAN) environments.

A test structure to separately measure sheet resistances of highly-doped-drain (HDD) and lightly-doped-drain (LDD) in LDD-type CMOSFETs with various gate spaces S having sub-100 nm sidewalls was proposed. From the reciprocal of source/drain-resistance R-1 versus S characteristics, the sheet resistance ρH of the high-conductive-region (HCR) corresponding to HDD and the approximate width WLC of the low-conductive-region (LCR) corresponding to LDD could be estimated. Both of ρH and WLC for p- and n-MOS devices were scarcely dependent on the gate voltage. The sidewall-width difference of 40 nm could be sufficiently detected by using the test structure with the S pitch of about 60 nm. The R-1 versus S characteristics showed the unstable resistance variations in the narrow S region less than 0.3 µm, which corresponded to the minimum S for the process used for the test device fabrication and suggested that various micro-loading effects seriously affected on the characteristics.

This paper studies a multiband mobile communication system to support both high data rate services and wide service coverage, using high and low frequency resources with different propagation characteristics. In the multiband system, multiple frequency bands are managed by a base station and one of the frequency bands is adaptively allocated to a terminal depending on his channel quality. By limiting the low frequency resources to a terminal not covered by the higher frequencies, the presented multiband system can accommodate many terminals providing wide coverage area, as if all radio resources have low frequency. From numerical results, the multiband system can provide wide service coverage area for much larger number of terminals than conventional systems. It is also found that an appropriate balance of multiple frequency resources is essential to achieve high capacity.