In this paper we consider blind source separation (BSS) problem of signals which are spatially uncorrelated of order four, but temporally correlated of order four (for instance speech or biomedical signals). For such type of signals we propose a new sufficient condition for separation using fourth order statistics, stating that the separation is possible, if the source signals have distinct normalized cumulant functions (depending on time delay). Using this condition we show that the BSS problem can be converted to a symmetric eigenvalue problem of a generalized cumulant matrix Z(4)(b) depending on L-dimensional parameter b, if this matrix has distinct eigenvalues. We prove that the set of parameters b which produce Z(4)(b) with distinct eigenvalues form an open subset of RL, whose complement has a measure zero. We propose a new separating algorithm which uses Jacobi's method for joint diagonalization of cumulant matrices depending on time delay. We empasize the following two features of this algorithm: 1) The optimal number of matrices for joint diago- nalization is 100-150 (established experimentally), which for large dimensional problems is much smaller than those of JADE; 2) It works well even if the signals from the above class are, additionally, white (of order two) with zero kurtosis (as shown by an example).

The present paper deals with the blind deconvolution of a Multiple-Input Multiple-Output Finite Impulse Response (MIMO-FIR) system. To deal with the blind deconvolution problem using the second-order statistics (SOS) of the outputs, Hua and Tugnait considered it under the conditions that a) the FIR system is irreducible and b) the input signals are spatially uncorrelated and have distinct power spectra. In the present paper, the problem is considered under a weaker condition than the condition a). Namely, we assume that c) the FIR system is equalizable by means of the SOS of the outputs. Under b) and c), we show that the system can be blindly identified up to a permutation, a scaling, and a delay using the SOS of the outputs. Moreover, based on this identifiability, we show a novel necessary and sufficiently condition for solving the blind deconvolution problem, and then, based on the condition, we propose a new algorithm for finding an equalizer using the SOS of the outputs, while Hua and Tugnait have not proposed any algorithm for solving the blind deconvolution under the conditions a) and b).

The square-law characteristics of MOSFET in the saturation region have a parameter of threshold voltage VT. However, it introduces some complexities to the circuit design since it depends on kinds of MOS technology and cannot be controlled easily. In this paper, we show an equivalent MOSFET cell which has VT-programming capability and some application instances based on it. The simulation is carried out using CMOS 0.8 µm n-well technology and the results have shown the feasibility of the proposed structure.

One purpose of Quality-of-Service (QoS) routing is to develop polynomial-time heuristic algorithms to tackle the MCP (multi-constrained-path) problem, which is NP-complete. In this paper, we introduce a new QoS routing heuristic framework, which focuses on how to increase the success ratio for finding a feasible path subject to multiple additive constraints. The key issue of this framework is to transform the single source single destination QoS routing problem to a single source multi-destination problem by expanding the destination vertex to its neighboring vertices. After that, the modified problem can be solved by existing source routing heuristic algorithms. The analysis and simulation results demonstrate that the framework can achieve a higher success ratio of finding a feasible path without increasing the computational complexity by setting the expansion operation properly.

A new source model for MoM analysis by using sinusoidal reaction matching technique is proposed for linear antenna analysis. This source model assumes a constant feeding gap and uniform electric field distribution inside the gap. The analysis results are compared with the results of the conventional models and measurement. It is found that the new model can incorporate the effect of the length of driving gap and is more accurate and more stable than that from the conventional source models. The proposed source model is simple and easy to use. This source model, together with the full kernel formulation, makes it possible to analyze the linear dipole antennas with no limitation of ratio of segment length to radius.

The radiation patterns are synthesized by properly disposing surface variations on dielectric rod waveguides. The genetic algorithm (GA) is applied for searching the optimum disposition of variation sections. A very fast calculation method used in the optimization is presented. The guided waves are related in the form of a 2-port circuit and the radiation field is expressed by superposition of the waves from variation sections. Various conical beams can be synthesized. Short variation sections and combination of several variation sections with different height are used to improve the synthesis performance. The ripple of the mainlobe and the sidelobe levels become small. Spherical sector patterns with a steep fall are synthesized and the agreement with the experimental values is confirmed.

This paper addresses a maximum likelihood method for source separation in the case of overdetermined mixtures corrupted by additive white Gaussian noise. We consider an approximate likelihood which is based on the Laplace approximation and develop a natural gradient adaptation algorithm to find a local maximum of the corresponding approximate likelihood. We present a detailed mathematical derivation of the algorithm using the Lie group invariance. Useful behavior of the algorithm is verified by numerical experiments.

A great deal of effort has been spent to develop strategies for allocation of resources in DS-CDMA systems in order to mitigate effects of interference between users. Here, the choice of spreading sequences and appropriate power allocation play a crucial role. When developing such strategies, CDMA system designers need to ensure that each user meets its quality-of-service requirement expressed in terms of the signal-to-interference+noise ratio. We say that a set of users is admissible in a CDMA system if one can assign sequences to the users and control their power so that all users meet their quality-of-service requirements. In [1], the problem of admissibility in a synchronous CDMA channel was solved. However, since the simplistic setting of perfect symbol synchronism rarely holds in practice, there is a strong need for investigating asynchronous CDMA channels. In this paper, we consider a K-user asynchronous CDMA channel with processing gain N and identical performance requirements for all users assuming chip synchronism. We solve the problem of admissibility of the users in such a channel if N K, and identify optimal sequences. We also show that constant power allocation is optimal. Results obtained in this paper give valuable insights into the limits of asynchronous CDMA systems.

A number of results on the estimation of direction of arrival have been obtained based on the assumption that the signal sources are point sources. Recently, it has been shown that signal source localization can be accomplished more adequately with distributed source models in some real surroundings. In this paper, we consider modeling of three-dimensional distributed signal sources, in which a source location is represented by the center angles and degrees of dispersion. We address estimation of the elevation and azimuth angles of distributed sources based on the proposed distributed source modeling in the three-dimensional space using two linear arrays. Some examples are included to more explicitly show the estimation procedures under the model: numerical results obtained by a MUSIC-based method with two uniform linear arrays are discussed.

In this paper, we propose an efficient quality-providing scheme to satisfy delay bound and loss ratio requirements for real-time video applications. To utilize network resources more efficiently while meeting service requirements, the network resources are dynamically allocated to each video connection based on the predicted traffic and currently provided quality of service degree. With the proposed bandwidth allocation method, a fair quality of service support in terms of packet loss ratio and maximum packet transfer delay to each video source can be achieved. To avoid possible quality violation by incoming new video connections, we present a connection admission control based on the provided QoS for existing connections and the measured traffic statistics. Simulation results show that our proposed dynamic method is able to provide accurate quality control.

This paper describes an automatic interface insertion scheme for in-system verification of algorithm models. To insert the interface, an algorithm model described in C is translated into another source code that includes the communication with hardware components in the target system to be validated with the algorithm model. The communication between the algorithm model and hardware components is achieved using transactors that perform transformation between access operations and bus cycle transactions. I/O terminal is introduced as an interface model to relate the transactions to access operations during the execution of the algorithm model, i.e., accesses to I/O terminals invoke bus cycle transactions in hardware and vice versa. An automatic interface insertion tool is developed using the source-to-source translation to identify the I/O terminals and insert interface function calls in the source code. The proposed automatic interface insertion scheme is validated by emulating several multimedia algorithms written in C on real target systems.

A problem of obtaining an optimal file transfer of a file transmission net N is to consider how to transmit, with the minimum total cost, copies of a certain file of information from some vertices, called sources, to other vertices of N by the respective vertices' copy demand numbers. This problem is NP-hard for a general file transmission net N. Some classes of N, on each of which a polynomial time algorithm for obtaining an optimal file transfer can be designed, are known. In the characterization, we assumed that file given originally to the source remains at the source without being transmitted. In this paper, we relax the assumption to the one that a sufficient number of copies of the file are given to the source and those copies can be transmitted from the source to other vertices on N. Under this new assumption, we characterize a class of file transmission nets, on each of which a polynomial time algorithm for obtaining an optimal file transfer can be designed. A minimum spanning tree with degree constraints plays a key role in the algorithm.

This paper discusses encoding of vehicular position information using predictive algorithms in inter-vehicle communications (IVC) from the viewpoints of source coding and noisy channels. Two vehicular driving models are assumed; one is the 15-mode as a suburban rapid transit driving pattern, the other is called calming mode as a street-driving pattern. Three types of schemes are compared; a pulse code modulation (PCM) scheme, a predictive coding (PC) scheme, and the variable interval prediction (VIP) scheme that is proposed here. This paper assumes that precise position information is got from a positioning system, and that all the transmitters and receivers have common predictors. Performance comparisons of the three types of schemes are carried out both of noiseless and noisy channels. Results show that the VIP scheme is superior to any other scheme.

In this paper, we propose a new resource management scheme for multiple video stream transmission in a wireless environment. The proposed scheme adaptively determines transmission parameters such as the number of assigned time slots, modulation format, and forward error correction (FEC) code rate according to the required bit rate and error sensitivity of the video stream as well as the channel state. The computer simulation results show that the proposed scheme drastically improves the image quality degradation due to channel errors.

This paper presents a new operation and maintenance system for PON (Passive Optical Network) subscriber loops and an assessment of its feasibility through experiments. This system adopts a broadband continuous wave light source as a testing signal and a fiber grating filter as a reflective component. The fiber grating filter is inserted somewhere in each subscriber loop for reflection of the testing signal. Our results show that the proposed system operates well for the purposes of quick decision of loop state and easy testing for PON subscriber loops.

This letter proposes a packet length-based group-wise transmission (LGT) rate scheduling scheme for non-real time data service for the uplink of direct sequence code division multiple access (DS/CDMA) system using the burst switching scheme to support the integrated voice/data service. The LGT scheme optimally determines two different rate groups and their optimal data rates so as to minimize the average packet transmission delay. It has shown that the packet transmission delay performance can be significantly improved over the conventional single-rate packet transmission scheme for integrated voice/data service. Furthermore, a main feature of the proposed scheme is simplicity in its implementation.

In this paper, we implemented practical resource adaptation mechanism for broadband application environment especially suitable for portable computers (note PCs) with limited power supply. Continuous development of portability enhancement and increasing computation with less power consumption for battery operating environment in note PC equips with intelligent resource controlling mechanism embedded inside the hardware. In such environment, where occasional environmental changes take in place rapidly, adaptive and collaborative controls of resources are required. Optimization of resource based on power supply management is dedicated to the operating environment. Bi-directional, user, application, operating system and device aware interface for resource configuration/management is extended to the current Unix operating system by implementing application programming interface of Advanced Configuration and Power Interface (ACPI). Traditional operating system conceals the active device probing and controlling method autonomously. Implementation of "State handler daemon" interface enables applications and application users to monitor and adapt the resources available during the operation. Thick collaboration between application and devices that are reserved in its limited environments economizes the consuming device utilization resulting: for example, life extension of battery life and effective network bandwidth adaptation. We focused on DVTS (Digital Video Transport System) as a typical broadband application. Evaluation shows 125 minutes of continuous battery operation with battery life extension compared to 70 minutes with traditional systems. Collaboration with network device management enables 100% packet transfer compared to 89% without any Resource adaptations.

A sound localization method based on the adaptive estimation of inverse Ear Canal Transfer Functions (ECTFs) using a stereo earphone-microphone combination is proposed. This method can adaptively obtain the individual's transfer functions to fit the listener in real-time. We evaluate our sound localization method by studying the relationship between the estimation error of inverse ECTFs and the auditory sound localization score perceived by several listener. As a result, we clarified that the estimation error required of inverse ECTFs are less than -10 dB. In addition, we describe two adaptive inverse filtering methods in order to realize real-time signal processing implementation using affine projection algorithm and discusses the convergence time of an adaptive inverse filter to determine the initial value. It is clarified that method 2 based on copy weights with initial value is more effective than method 1 with filtered-x algorithm, in terms of convergence, if the initial value is the average of many listeners' impulse responses for our sound localization method.

Wyner and Ziv considered the rate-distortion function for source coding with side information at the decoder (we call the Wyner-Ziv problem). In this paper we show an information-spectrum approach to the Wyner-Ziv problem for general class of nonstationary and/or nonergodic sources with side information at the decoder, where the distortion measure is arbitrary and may be nonadditive. We show that a general formula for the rate-distortion function of the Wyner-Ziv problem for general sources with the maximum distortion criterion under fixed-length coding by using the information spectrum approach.

This paper describes a method for hands-free speech recognition, and particularly for the simultaneous recognition of multiple sound sources. The method is based on the 3-D Viterbi search, i.e., extended to the 3-D N-best search method enabling the recognition of multiple sound sources. The baseline system integrates two existing technologies--3-D Viterbi search and conventional N-best search--into a complete system. Previously, the first evaluation of the 3-D N-best search-based system showed that new ideas are necessary to develop a system for the simultaneous recognition of multiple sound sources. It found two factors that play important roles in the performance of the system, namely the different likelihood ranges of the sound sources and the direction-based separation of the hypotheses. In order to solve these problems, we implemented a likelihood normalization and a path distance-based clustering technique into the baseline 3-D N-best search-based system. The performance of our system was evaluated through experiments on simulated data for the case of two talkers. The experiments showed significant improvements by implementing the above two techniques. The best results were obtained by implementing the two techniques and using a microphone array composed of 32 channels. More specifically, the Word Accuracy for the two talkers was higher than 80% and the Simultaneous Word Accuracy (where both sources are correctly recognized simultaneously) was higher than 70%, which are very promising results.