In forthcoming home network environment, computation capability will be embedded invisibly in home appliances, sensors, walls, ceilings, and floors. People will conduct various tasks using multiple devices simultaneously without consciousness of using computers. In this paper, first, we propose an application model named Virtual Network Appliance (VNA) model which simplify and expand device utilization. In the model, each device has VNA runtime system and function objects, called VNA components, running on it. A user task is defined in an application called VNA which is a logical appliance consisting of abstract function requirements and a message graph among them. Second, we propose Virtual Plug&Play mechanism which is a dynamic service integration mechanism in VNA model implementation. When a user conducts a task, he/she makes a VNA runtime system on a user-side terminal load a VNA definition appropriate for the task. Virtual Plug&Play dynamically discovers required VNA components and establishes the message graph as defined. Since XML documents are used to describe a VNA, users can share and customize it easily. We call the device integration done by Virtual Plug&Play top-down integration, which existing middleware do not aim at. Finally, we show that Virtual Plug&Play affords practical performance for top-down integration by performance evaluation.

This letter shows the results of a series of link level simulations conducted to evaluate the performances of spatial and temporal equalizers (S/T-equalizers) using field measurement data. The configuration of the spatial and temporal equalizer discussed in this letter can be expressed as a cascade of an adaptive array antenna and maximum likelihood sequence estimator (MLSE): each of the adaptive array antenna elements has a fractionally spaced tapped delay line (FTDL), and the MLSE has taps covering a portion of channel delay profile. Bit error rate (BER) performances of the S/T-equalizers are presented, and performance sensitivity to symbol timing offset is investigated.

Providing results of a series of link-level simulations for a class of spatial and temporal equalizer (S/T-equalizer) is the primary objective of this letter, which is supplemental to this letter's companion article. The S/T-equalizers discussed in this letter have a configuration that can be expressed as the cascaded connection of adaptive array antenna and maximum likelihood sequence estimator (MLSE): each of the adaptive array antenna elements has a fractionally spaced tapped delay line (FTDL), and the MLSE has taps covering a portion of the channel delay profile. Both the desired and interference signals suffer from severe inter-symbol interference (ISI). A major difference of this article from its companion letter is that account is taken of the presence of co-channel interference (CCI). Bit error rate (BER) performance of the S/T-equalizer is presented as a result of the link-level simulations that use field measurement data.

An important goal in communication theory is to construct good minimum squared Euclidean distance (MSED) codes for transmission over additive white Gaussian noise (AWGN) channels. In this paper, a new construction method for the M-ary phase-shift-keyed (M-PSK) codes over the ring structure ZM, the ring of integers modulo M, with a good minimum Euclidean distance, is proposed. The proposed codes are linear when multiple coset leaders are considered. The characteristics and performance levels of the newly constructed codes are analyzed for code length up to n 8. It is found that the proposed codes compare favorably with Piret's codes and Graeffe's method codes on Gaussian channels in terms of decoding complexity, coding gain, and error performance.

Adaptation of software components to the requirements is one of the key concerns in Component Based Software Development (CBSD). In this paper, we propose a formal approach to compose component based systems which are adaptable to the requirements. We focus on the functional aspects of software components and requirements, which are expressed in S-sorted functions. Those S-sorted functions are transformed into Colored Petri Nets (CPN) models in order to evaluate connectivity between the components, and to evaluate adaptability of composed systems to the requirements. The connectivity is measured based on colors or data types in CPN, while the adaptability is measured based on functional equivalency. We introduce simple glue codes to connect the components each other. The paper focuses on business applications, however the proposed approach can be applied to any other domains as far as the functional adaptability is concerned.

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.

This paper presents the outline of the systolic array recursive least-squares (RLS) processor prototyped primarily with the aim of broadband mobile communication applications. To execute the RLS algorithm effectively, this processor uses an orthogonal triangularization technique known in matrix algebra as QR decomposition for parallel pipelined processing. The processor board comprises 19 application-specific integrated circuit chips, each with approximately one million gates. Thirty-two bit fixed-point signal processing takes place in the processor, with which one cycle of internal cell signal processing requires approximately 500 nsec, and boundary cell signal processing requires approximately 80 nsec. The processor board can estimate up to 10 parameters. It takes approximately 35 µs to estimate 10 parameters using 41 known symbols. To evaluate signal processing performance of the prototyped systolic array processor board, processing time required to estimate a certain number of parameters using the prototyped board was comapred with using a digital signal processing (DSP) board. The DSP board performed a standard form of the RLS algorithm. Additionally, we conducted minimum mean-squared error adaptive array in-lab experiments using a complex baseband fading/array response simulator. In terms of parameter estimation accuracy, the processor is found to produce virtually the same results as a conventional software engine using floating-point operations.

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 a mechanism to facilitate the development of component-based mobile applications with adaptive behaviors. The design principles and communication patterns of legacy software systems will greatly change in a forthcoming environment, where a variety of computing devices become embedded in home and office environments, users move around with/without portable computing devices, and all the devices are interconnected through wired/wireless networks. In the proposed mechanism, Improvised Assembly Mechanism (IAM), we realize functionality to compose an application in an ad hoc manner and to achieve the adaptation of applications by adding, replacing, supplementing, and relocating components at system runtime according to various environmental changes such as the locational changes of computing devices and users. The mechanism is implemented as a built-in functionality of the Soul component, which is one of the fundamental elements in the Possession model.

An improved pulse shaper is proposed which is able to control both the spatial and temporal profile of femtosecond light pulses. Our pulse shaper exploits the spatio-temporal coupling effect seen in pulse shapers. Its properties are numerically analyzed by application of the Wigner distribution function. We confirm that the spatio-temporal output pulse track dictates the differentiation of the phase mask; that the degree of spatio-temporal coupling is determined by the focal length ratio of the lenses in the pulse shaper; and that space to spatial-frequency chirp results from misalignment of lenses.

A proper design and analysis of future wideband wireless communication systems require an accurate radio channel model. This model is claimed to characterize both the spatial and temporal channel characteristics. This paper investigates the spatio-temporal channel modeling based on a ray-tracing approach. The temporal channels are characterized by a delay profile. The statistical median and fading-fluctuation range of delay profiles are predicted from ray tracing by incorporating the random phase approach. A high level of agreement between predicted results and measured ones is observed in the verification. The spatio-temporal channel impulse response (CIR) predicted from ray tracing is also transformed to have limited band-width and limited beam-width characteristics. The applicability of this transformation is also verified by the comparison with measurement. These verifications prepare the ground for the use of ray-tracing approaches to evaluate system performance in real environments.

First, we show that the necessary and sufficient condition for the existence of a balanced bowtie decomposition of the complete tripartite multi-graph λ Kn1,n2,n3 is (i) n1=n2=n3 0 (mod 6) for λ 1,5 (mod 6), (ii) n1=n2=n3 0 (mod 3) for λ 2,4 (mod 6), (iii) n1=n2=n3 0 (mod 2) for λ 3 (mod 6), and (iv) n1=n2=n3 2 for λ 0 (mod 6). Next, we show that the necessary and sufficient condition for the existence of a balanced trefoil decomposition of the complete tripartite multi-graph λ Kn1,n2,n3 is (i) n1=n2=n3 0 (mod 9) for λ 1,2,4,5,7,8 (mod 9), (ii) n1=n2=n3 0 (mod 3) for λ 3,6 (mod 9), and (iii) n1=n2=n3 3 for λ 0 (mod 9).

Recently, two approaches investigated indexing and retrieving videos. One approach utilized the visual features of individual objects, and the other approach exploited the spatio-temporal relationships between multiple objects. In this paper, we integrate both approaches into a new video model, called the Visual-Spatio-Temporal (VST) model to represent videos. The visual features are modeled in a topological approach and integrated with the spatio-temporal relationships. As a result, we defined rich sets of VST relationships which support and simplify the formulation of more semantical queries. An intuitive query interface which allows users to describe VST features of video objects by sketch and feature specification is presented. The conducted experiments prove the effectiveness of modeling and querying videos by the visual features of individual objects and the VST relationships between multiple objects.

Spatio-temporal decorrelation is the task of eliminating correlations between associated signals in spatial domain as well as in time domain. In this paper, we present a simple but efficient adaptive algorithm for spatio-temporal decorrelation. For the task of spatio-temporal decorrelation, we consider a dynamic recurrent network and calculate the associated natural gradient for the minimization of an appropriate optimization function. The natural gradient based spatio-temporal decorrelation algorithm is applied to the task of blind deconvolution of linear single input multiple output (SIMO) system and its performance is compared to the spatio-temporal anti-Hebbian learning rule.

In this paper, we propose fast bitmap search algorithms to reduce the computational complexity of transform-based vector quantization (VQ) techniques, which achieve better quality in reconstructed images than the ordinary VQ. By removing the unlikely codewords in each step, the bitmap search method, which starts from the most significant bitmap then the successive significant ones, can save more than 90% computation of the ordinary transformed VQ. By applying to the singular value decomposition (SVD) VQ as an example, theoretical analyses and simulation results show that the proposed bitmap search methods dramatically reduce the computation and achieve invisible distortion in the reconstructed images.

This paper presents a method of thread composition in a hardware compiler Bach. Bach synthesizes RT level circuits from a system description written in Bach-C language, where a system is modeled as communicating processes running in parallel. The system description is decomposed into threads, i.e., strings of sequential processes, by grouping processes which are not executed in parallel. The set of threads are then converted into behavioral VHDL models and passed to a behavioral synthesizer. The proposed method attempts to find a thread configuration that maximize resource sharing among processes in the threads. Experiments on two real designs show that the circuit sizes were reduced by 3.7% and 14.7%. We also show the detailed statistics and analysis of the size of the resulting gate level circuits.

In computer science, concepts of resource such as data consumption and of time such as execution time are very important. Logical systems which can treat them have been applied in that field. Linear logic has been called a resource conscious logic. The expressive power is enough to describe a dynamic change in process environments. However, linear logic is not enough to treat a dynamic change in environments with the passage of time since it does not include a concept of time directly. A typical example is the relation between linear logic and Petri nets. It is well known that the reachability problem for Petri nets is equivalent to the provability for the corresponding sequent of linear logic. But linear logic cannot naturally represent timed Petri nets which are extensions of ordinary Petri nets with respect to time concept. So we extend linear logic with respect to time concept in order to introduce a resource-conscious and time-dependent logical system, that is, temporal linear logic. This system has some temporal operators "" which means a resource usable only once at the next time, "" which means a resource usable only once at anytime, and a modal storage operator "!" which means a resource usable any times at anytime. We can show that the reachability problem for timed Petri nets is equivalent to the provability for the corresponding sequent of temporal linear logic. In this paper, we also represent the description of synchronous communication model by temporal linear logic. The expressive power of temporal linear logic will be applicable to various fields of computer science.

To estimate moving object velocity in an image sequence is useful for a variety of applications, such as velocity measurement, computer vision and monitoring systems. An effective way is to approach it in the transform/spatiotemporal mixed domain (MixeD), which transforms the 3-D signal processing problem into 1-D complex signal processing. But it remains a problem how to select several spatial frequency points in the MixeD which may influence the accuracy of velocity estimation and object detection. In this paper, a spatial frequency selection method has been proposed, which can choose the appropriate spatial frequency points out of a number of points in MixeD automatically. So the velocity estimation problem can be addressed by solving the coupled equations established over two selected appropriate points in 2-D spatial frequency domain other than searching for the spectral energy plane over a number of points selected by experience. In this method, evaluation functions corresponding to image sequence with one moving object and two moving objects are established firstly, and the selection is then achieved by using the established evaluation functions together with a threshold. The simulation results show that the proposed method is effective on the appropriate spatial frequency selection.

We present a design method of the simulation probability density function for a trellis-coded modulation (TCM) in an impulsive noise environment. The upper bound evaluation method for the TCM scheme cannot be applied to the lognormally distributed impulsive noise, since the Chernoff bound cannot be defined. Thus the error probability can only be estimated by a computer simulation. For an evaluation of a low error probability, importance sampling (IS) is an efficient technique. A design method of the simulation probability density function, which plays an important role in IS, is proposed for the noise. The effectivity is shown by a numerical example.

This paper presents a three-dimensional polarimetric detection result of targets buried in snowpack by synthetic aperture FM-CW radar system. Since the FM-CW radar is suitable for short range sensing and can be equipped with fully polarimetric capability, we further extended it to a polarimetric three-dimensional SAR system. A field experiment was carried out to image and detect targets in a natural snowpack of 280 cm deep. The polarimetric detection and identification schemes are the polarimetric filtering, three-component decomposition, and the power polarization anisotropy coefficient. These approaches to acquired data show the usefulness of three-dimensional polarimetric FM-CW SAR system.