Resilient Packet Ring (RPR) is a new technology currently being standardized in the IEEE 802.17 working group. The existed bandwidth allocation algorithms for RPR networks are not able to provide satisfactory solutions to meet the performance requirements. In this paper we propose one fair bandwidth allocation algorithm, termed PID-RPR, which satisfies the performance goals of RPR networks, such as fairness, high utilization and maximal spatial reuse. The algorithm is operated at each RPR node in a distributive way; the proportional, integral and differential (PID) controller is used to allocate bandwidth on the outgoing link of the node for the flows over the link in a weighted manner. To achieve the global coordination, one control packet containing every node's message runs around the ring in order to update the relevant message for all nodes on the ring. When the packet reaches one node, this node adjusts its own rate according to its own message in the control packet; in the meantime it updates other nodes' control message in the control packet. As the control packet propagates around the ring, each node can eventually adjust its sending rate to reach its fair share according to the fairness criterion, and the buffer occupancy at each node is kept within the target value. Our algorithm is of distributed nature in the sense that upstream ring nodes inject traffic at a rate according to congestion and fairness criteria downstream. The simulation results demonstrate that satisfactory performance of RPR networks can be achieved under the proposed bandwidth allocation scheme.

In this paper, the 1-D real-valued discrete Gabor transform (RDGT) proposed in our previous work and its relationship with the complex-valued discrete Gabor transform (CDGT) are briefly reviewed. Block time-recursive RDGT algorithms for the efficient and fast computation of the 1-D RDGT coefficients and for the fast reconstruction of the original signal from the coefficients are then developed in both the critical sampling case and the oversampling case. Unified parallel lattice structures for the implementation of the algorithms are studied. And the computational complexity analysis and comparison show that the proposed algorithms provide a more efficient and faster approach for the computation of the discrete Gabor transforms.

This paper presents a framework for automatic video region-of-interest determination based on visual attention model. We view this work as a preliminary step towards the solution of high-level semantic video analysis. Facing such a challenging issue, in this work, a set of attempts on using video attention features and knowledge of computational media aesthetics are made. The three types of visual attention features we used are intensity, color, and motion. Referring to aesthetic principles, these features are combined according to camera motion types on the basis of a new proposed video analysis unit, frame-segment. We conduct subjective experiments on several kinds of video data and demonstrate the effectiveness of the proposed framework.

This paper proposes per-tone equalization methods for single carrier block transmission with cyclic prefix (SC-CP) systems. Minimum mean-square-error (MMSE) based optimum weights of the per-tone equalizers are derived for SISO (single-input single-output), SIMO (single-input multiple-output), and MIMO (multiple-input multiple-output) SC-CP systems. Unlike conventional frequency domain equalization methods, where discrete Fourier transform (DFT) is employed, the per-tone equalizers utilize sliding DFT, which makes it possible to achieve good performance even when the length of the guard interval is shorter than the channel order. Computer simulation results show that the proposed equalizers can significantly improve the bit error rate (BER) performance of the SISO, SIMO, and MIMO SC-CP systems with the insufficient guard interval.

A new and simple image processing approach for the measurement of the length of pedestrian crossings with a view to develop a travel aid for the blind people is described. In a crossing, the usual black road surface is painted with constant width periodic white bands. The crossing length is estimated using vector geometry from the left- and the right-border lines, the first-, the second- and the end-edge lines of the crossing region. Image processing techniques are applied on the crossing image to find these lines. Experimental results using real road scenes with pedestrian crossing confirm the effectiveness of the proposed method.

In this paper, we propose a new one-dimensional (1D) integer discrete cosine transform (Int-DCT) for unified lossless/lossy image compression. The proposed 1D Int-DCT is newly designed to reduce rounding effects by minimizing number of rounding operations. The proposed Int-DCT can be operated not only lossless coding for a high quality decoded image but also lossy coding for a compatibility with the conventional DCT-based coding system. Both theoretical analysis and simulation results confirm an effectiveness of the proposed Int-DCT.

This paper presents a VLSI processor for high-speed and reliable stereo matching based on adaptive window-size control of SAD(Sum of Absolute Differences) computation. To reduce its computational complexity, SADs are computed using multi-resolution images. Parallel memory access is essential for highly parallel image processing. For parallel memory access, this paper also presents an optimal memory allocation that minimizes the hardware amount under the condition of parallel memory access at specified resolutions.

A seven-port wave-correlator based vector network analyzer is proposed. The seven-port wave-correlator is a combination of two six-port wave-correlators which share common components. Furthermore, the complex wave ratio measurement accuracy is improved since the input signals can be directly detected by the side-arm ports. A seven-port wave-correlator is fabricated using microstrip branch line couplers. The performance of the wave-correlator and the constructed network analyzer are evaluated, and the measurement accuracy is confirmed.

Our goal in this paper is to provide a complete detection analysis for the OS processor along with OSGO and OSSO modified versions, for M postdetection integrated pulses when the operating environment is nonideal. Analytical results of performance are presented in both multiple-target situations and in regions of clutter power transitions. The primary and the secondary interfering targets are assumed to be fluctuating in accordance with the Swerling II target fluctuation model. As the number of noncoherently integrated pulses increases, lower threshold values and consequently better detection performances are obtained in both homogeneous and multiple target background models. However, the false alarm rate performance of OSSO-CFAR scheme at clutter edges is worsen with increasing the postdetection integrated pulses. As predicted, the OSGO-CFAR detector accommodates the presence of spurious targets in the reference window, given that their number is within its allowable range in each local window, and controls the rate of false alarm when the contents of the reference cells have clutter boundaries. The OSSO-CFAR scheme is useful in the situation where there is a cluster of radar targets amongst the estimation cells.

Digital Terrestrial Television (DTV) services began in Japan in December 2003. This paper proposes a novel on-glass antenna for mobile reception of terrestrial television. The gain of the proposed antenna is 4.7 dB higher than commercial monopole antennas when installed on a vehicle. Other merits of this antenna are a broad input impedance bandwidth across the UHF band (470-710 MHz), and the fact that it does not spoil vehicle appearance. Field experiments have confirmed that a diversity system using four of the proposed antennas is capable of mobile DTV reception.

In this letter, we propose a 2-D receiver structure for multicarrier code division multiple access (MC-CDMA) systems with the reduced-rank multistage Wiener filter. Due to the fast convergence property of the reduced-rank processing, it outperforms MMSE-based receivers with the classical Wiener solution, which is estimated by using a limited number of samples.

This letter compares orthogonal space time codes and quasi-orthogonal codes when the wireless channels are fast fading. It is well known that a orthogonal space-time code is better than a quasi-orthogonal code in high signal-to-noise ratio (SNR) range and that a quasi-orthogonal code is better in low SNR range. In this letter, we show that a quasi-orthogonal space-time code is a better choice even in high SNR range when the channels are fast fading.

This paper deals with the decomposition of surface data into several fractal signal based on the parameter estimation by the Mean Likelihood and Importance Sampling (IS) based on the Monte Carlo simulations. The method is applied to the feature extraction of surface data. Assuming the stochastic models for generating the surface, the likelihood function is defined by using wavelet coefficients and the parameter are estimated based on the mean likelihood by using the IS. The approximation of the wavelet coefficients is used for estimation as well as the statistics defined for the variances of wavelet coefficients, and the likelihood function is modified by the approximation. After completing the decomposition of underlying surface data into several fractal surface, the prediction method for the fractal signal is employed based on the scale expansion based on the self-similarity of fractal geometry. After discussing the effect of additive noise, the method is applied to the feature extraction of real distribution of surface data such as the cloud and earthquakes.

This paper presents a study on the blind separation of a convoluted mixture of speech signals using Frequency Domain Independent Component Analysis (FDICA) algorithm based on the negentropy maximization of Time Frequency Series of Speech (TFSS). The comparative studies on the negentropy approximation of TFSS using generalized Higher Order Statistics (HOS) of different nonquadratic, nonlinear functions are presented. A new nonlinear function based on the statistical modeling of TFSS by exponential power functions has also been proposed. The estimation of standard error and bias, obtained using the sequential delete-one jackknifing method, in the approximation of negentropy of TFSS by different nonlinear functions along with their signal separation performance indicate the superlative power of the exponential-power-based nonlinear function. The proposed nonlinear function has been found to speed-up convergence with slight improvement in the separation quality under reverberant conditions.

In an Orthogonal Frequency Division Multiplexing (OFDM) systems, the Peak to Average power Ratio (PAR) is high. The clipping signal scheme is a useful and simple method to reduce the PAR. However, it introduces additional noise that degrades the systems performance. We propose an oversampling scheme to deal with the received signal in order to reduce the clipping noise by using finite impulse response (FIR) filter. Coefficients of the filter are obtained by correlation function of the received signal and the oversampling information at receiver. The performance of the proposed technique is evaluated for frequency selective channel. Results show that the proposed scheme can mitigate the clipping noise significantly for OFDM systems and in order to maintain the system's capacity, the clipping ratio should be larger than 2.5.

A robust microphone array for speech enhancement and noise suppression is studied in this paper. To overcome target signal cancellation problem of conventional beamformer caused by array imperfections or reverberation effects of acoustic enclosure, the proposed microphone array adopts an arbitrary model of channel transfer function (TF) relating microphone and speech source. Since the estimation of channel TF itself is often intractable, herein, transfer function ratio (TFR) is estimated instead and used to form a suboptimal beamformer. A robust TFR estimation method is proposed based on signal subspace analysis technique against stationary or slowly varying noise. Experiments using simulated signal and actual signal recorded in a real room illustrate that the proposed method has high performance in adverse environment.

In a CDMA non-linear interference canceller, a generated replica of an interference signal is multiplied by a positive number smaller than unity, which is called cancellation moderating factor (CMF), to prevent interference enhancement due to inaccurate replica subtraction. In this paper, two CMF controlling schemes applicable to a multistage parallel interference canceller with multi-antenna (spatial diversity) reception are proposed. They control CMF by using the mean square error of the complex channel gain or by using the ratio of the estimated power of each interference signal to remaining interference signals' power, in order to mitigate the replica subtraction error due to inaccurate channel estimation. The performance of the proposed schemes are evaluated by computer simulations assuming an asynchronous uplink single chip-rate variable spreading factor DS-CDMA system. The simulation results show that the proposed schemes with higher order diversity reception improve the bit error rate (BER) performance compared with a conventional scheme considering the tentative decision error or fixed CMF settings. Their performance improvement is by 0.1-0.9 dB in terms of the required Eb/N0 at an average BER of 10-5 over exponentially decaying 5-path Rayleigh distributed channels when the number of receiving antennas is 6.

This paper introduces the multiple signal classification (MUSIC) method that utilizes the transfer characteristics of microphones located at the same place, namely aggregated microphones. The conventional microphone array realizes a sound localization system according to the differences in the arrival time, phase shift, and the level of the sound wave among each microphone. Therefore, it is difficult to miniaturize the microphone array. The objective of our research is to build a reliable miniaturized sound localization system using aggregated microphones. In this paper, we describe a sound system with N microphones. We then show that the microphone array system and the proposed aggregated microphone system can be described in the same framework. We apply the multiple signal classification to the method that utilizes the transfer characteristics of the microphones placed at a same location and compare the proposed method with the microphone array. In the proposed method, all microphones are placed at the same place. Hence, it is easy to miniaturize the system. This feature is considered to be useful for practical applications. The experimental results obtained in an ordinary room are shown to verify the validity of the measurement.

This paper focuses on the underdetermined blind source separation (BSS) of three speech signals mixed in a real environment from measurements provided by two sensors. To date, solutions to the underdetermined BSS problem have mainly been based on the assumption that the speech signals are sufficiently sparse. They involve designing binary masks that extract signals at time-frequency points where only one signal was assumed to exist. The major issue encountered in previous work relates to the occurrence of distortion, which affects a separated signal with loud musical noise. To overcome this problem, we propose combining sparseness with the use of an estimated mixing matrix. First, we use a geometrical approach to detect when only one source is active and to perform a preliminary separation with a time-frequency mask. This information is then used to estimate the mixing matrix, which allows us to improve our separation. Experimental results show that this combination of time-frequency mask and mixing matrix estimation provides separated signals of better quality (less distortion, less musical noise) than those extracted without using the estimated mixing matrix in reverberant conditions where the reverberant time (TR) was 130 ms and 200 ms. Furthermore, informal listening tests clearly show that musical noise is deeply lowered by the proposed method comparatively to the classical approaches.

Call admission control (CAC) plays a significant role in providing the efficient use of the limited bandwidth and the desired quality-of-service (QoS) in mobile multimedia communications. As efficiency is an important performance issue for CAC in the mobile networks with multimedia services, the concept of fairness among services should also be considered. Game theory provides an appropriate framework for formulating such fair and efficient CAC problem. Thus, in this paper, a framework based on game theory (both of noncooperative and cooperative games) is proposed to select fair-efficient guard bandwidth coefficients of the CAC scheme for the asymmetrical traffic case in mobile multimedia communications. The proposed game theoretic framework provides fairness and efficiency in the aspects of bandwidth utilization and QoS for multiple classes of traffic, and also guarantees the proper priority mechanism. Call classes are viewed as the players of a game. Utility function of the player is defined to be of two types, the bandwidth utilization and the weighted sum of new call accepting probability and handoff succeeding probability. The numerical results show that, for both types of the utility function, there is a unique equilibrium point of the noncooperative game for any given offered load. For the cooperative game, the arbitration schemes for the interpersonal comparisons of utility and the bargaining problem are investigated. The results also indicate that, for both types of the utility function, the Nash solution with the origin (0,0) as the starting point of the bargaining problem can achieve higher total utility than the previous CAC scheme while at the same time providing fairness by satisfying a set of fairness axioms. Since the Nash solution is determined from the domain of the Pareto boundary, the way to generate the Pareto boundary is also provided. Therefore, the Nash solution can be obtained easily.