One of the main issues of Mobile IPv6 is handover latency that causes service disruption time. Although plenty of proposals significantly reduce the service disruption time, they suffer from redundant routing that causes packet misordering and bandwidth consumption during the process of inter-domain handover. In this paper, we propose a new scheme that minimizes the redundant routing during the process of inter-domain handover by utilizing forwarding routers for each correspondent node. Our proposed scheme consists of forwarding router discovery and proactive handover. We evaluate our proposed scheme in the view of packet misordering and bandwidth consumption, and clarify the efficiency of our proposed scheme. We also evaluate the impact of the forwarding routers' capacity since routers have limited resources. By strategically locating forwarding routers, e.g. next to the router that has peering to another domain, the redundant routing caused by inter-domain handover will be efficiently suppressed.

While mobile networks have been enhanced to support a variety of mobile multimedia services such as video telephony and rich data content delivery, a new challenge is being created by the remarkable development of micro-device technologies such as micro processor-chips, sensors, and RF tags. These developments suggest the rapid emergence of the ubiquitous computing environment; computers supporting human life without imposing any stress on the users. The combination of broadband global networks and ubiquitous computing environment will lead to an entirely new class of services, which we call ubiquitous networking services. This paper discusses how to create ubiquitous service environments comparing global networking approaches which are based on fixed and mobile networks. It is shown that the mobile approach is better from service applicability and reliability viewpoints. Networking architecture is proposed which expand 4G mobile cellular networks to real space via gateways on the edges of the mobile network (i.e. mobile terminals). A new set of technical requirements will emerge via this approach, which may accelerate the paradigm shift from the current mobile network architecture and even from the Internet of today.

This paper presents a novel low-power register file based on adiabatic logic. The register file consists of a storage-cell array, address decoders, read/write control circuits, sense amplifiers, and read/write drivers. The storage-cell array is based on the conventional memory cell. All the circuits except the storage-cell array employ CPAL (complementary pass-transistor adiabatic logic) to recover the charge of large node capacitance on address decoders, bit-lines and word-lines in fully adiabatic manner. The minimization of energy consumption was investigated by choosing the optimal size of CPAL circuits for large load capacitance. The power consumption of the proposed adiabatic register file is significantly reduced because the energy transferred to the large capacitance buses is mostly recovered. The energy and functional simulations are performed using the net-list extracted from the layout. HSPICE simulation results indicate that the proposed register file attains energy savings of 65% to 85% as compared to the conventional CMOS implementation for clock rates ranging from 25 to 200 MHz.

When video data are transmitted via the network, the quality of video data must be carefully chosen to be best under the condition that the transmission is not influenced by other internet services. They often use the simulcast type, which uses independent streams that are stored and transmitted for the quality, considering implementation, when they select the video quality. On the other hand, we had already proposed the scalable structure, which consists of base and enhancement data, but when they require the high quality video, these data are combined using the transcoding methods. In this paper, we propose the video contents delivery methods with scalable transcoding, in which users can update the quality of video data even after the transmission by base data and differential data. In order to reduce the total time of not only users' access time, but also watching time, we compare simulcast method with proposed methods in the total content utilization time using a video contents access model, and evaluate required transcoding time to reduce the waiting time of users.

This paper proposes "Multipath Control and Proactive Control" to realize a robust QoS control system for mobile multimedia communication in an IP-based cellular network. In this network, all kinds of traffic will share the same backbone network. This requires a QoS system that differentiates services according to the required quality. Though DiffServ is thought to be a promising technique for achieving QoS, an effective path control scheme and a technique that is suitable enough for rapid traffic changes are not yet available. Our solution is multipath control using linear optimization combined with proactive control using traffic anomaly detection. Simulation results show that multipath control and proactive control improve system performance in terms of throughput and packet loss when rapid traffic change takes place.

This paper presents an adaptive control design for the ABR traffic congestion control in ATM networks. Firstly, we consider a control-based mathematical model to the ABR traffic congestion control problem. Then the feedback pole placement control design is applied to the ATM ABR traffic congestion control problem for the case of known delays. Finally, by using the online plant parameter estimation algorithm and modifying the controller parameters adaptively in real time, a method to treat the case of unknown time-varying delays is proposed. Several design modifications are introduced to solve practical control issues such as bounded command rate constraint, output buffer saturation and bounded values to the plant parameter estimation algorithm. Simulations are implemented to verify the proposed control design. It is shown that while considering these practical control issues, the control method satisfies the requirements of fairness to users, network efficiency, unknown time-varying delays, queue length control and good convergence performance at an acceptable computation effort.

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.

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 order to assess the effect of multipath fading on Orthogonal Frequency Division Multiplex (OFDM) signal transmission when the delay profile exceeds the guard interval, a simple prediction model is developed based on the Equivalent Transmission-Path (ETP) model. This model, which is described in this paper, is referred to as the ETP-OFDM-statistical model. The validity of the model is demonstrated by comparing the calculated digital transmission characteristics to results obtained by computer simulation. Using the newly developed ETP-OFDM-statistical model, digital transmission characteristics of the OFDM signal in a multipath environment when the delay profile exceeds the guard interval are shown as a function of delay spread, guard interval and OFDM symbol period.

Taking the uplink and downlink cochannel interference and noise into account, we determine the error probability in detecting a binary phase-shift keying (BPSK) signal transmitted over a satellite system containing two high power amplifiers (HPA). The first one is the constituent part of the transmitting ground station and the second one is the constituent part of the satellite station. The emphasis is placed on determining the system performance degradation imposed by the influence of the nonlinear characteristic of the HPA at the transmitting ground station in combination with the negative influences of the uplink and downlink cochannel interference, as well as the nonlinear characteristic of HPA at the satellite station.

A speech signal captured by a distant microphone is generally smeared by reverberation, which severely degrades both the speech intelligibility and Automatic Speech Recognition (ASR) performance. Previously, we proposed a single-microphone dereverberation method, named "Harmonicity based dEReverBeration (HERB)." HERB estimates the inverse filter for an unknown room transfer function by utilizing an essential feature of speech, namely harmonic structure. In previous studies, improvements in speech intelligibility was shown solely with spectrograms, and improvements in ASR performance were simply confirmed by matched condition acoustic model. In this paper, we undertook a further investigation of HERB's potential as regards to the above two factors. First, we examined speech intelligibility by means of objective indices. As a result, we found that HERB is capable of improving the speech intelligibility to approximately that of clean speech. Second, since HERB alone could not improve the ASR performance sufficiently, we further analyzed the HERB mechanism with a view to achieving further improvements. Taking the analysis results into account, we proposed an appropriate ASR configuration and conducted experiments. Experimental results confirmed that, if HERB is used with an ASR adaptation scheme such as MLLR and a multicondition acoustic model, it is very effective for improving ASR performance even in unknown severely reverberant environments.

In this paper, we propose a new on-line adaptive relaxation algorithm for an inverse filter in a multichannel sound reproduction system. The fluctuation of room transfer functions degrades reproduced sound in conventional sound reproduction systems in which the coefficients of the inverse filter are fixed. In order to resolve this problem, an iterative relaxation algorithm for an inverse filter performed by truncated singular value decomposition (adaptive TSVD) has been proposed. However, it is difficult to apply this method within the time duration of the sound of speech or music in the original signals. Therefore, we extend adaptive TSVD to an on-line-type algorithm based on the observed signal at only one control point, normalizing the observed signal with the original sound. The result of the simulation using real environmental data reveals that the proposed method can always carry out the relaxation process against acoustic fluctuation, for any time duration. Also, subjective evaluation in the real acoustic environment indicates that the sound quality improves without degrading the localization.

As the number of cluster system users increases, it is important to maintain stable operation. Although hardware preventive maintenance is important for sustaining smooth operation, hardware testing tools for cluster systems have received little attention. In this paper, we propose a memory testing tool for Linux cluster systems.

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 cascode-pair distributed amplifier design approach using 0.25 µm GaAs-based PHEMT MMIC technology, which covers 2-32 GHz. Electromagnetic simulation results show that this amplifier achieves 18 dB gain from 2 to 32 GHz and 0.5 dB gain flatness over the band. The reflected coefficients at the input and output ports are below -10 dB up to 27 GHz. The output power at 1 dB compression is greater than 24 dBm at 20 GHz. An appropriate feedback resistance can be utilized to improve P1 dB for about 6 dBm. The DOE (design of experiment) approach is carried out by a simulation tool for better performance and tolerance of the devices is also analyzed. The circuit configuration is capable of operating over ultra-broad band amplification.

A rotator graph was proposed as a topology for interconnection networks of parallel computers, and it is promising because of its small diameter and small degree. However, a rotator graph is a directed graph that sometimes behaves harmfully when it is applied to actual problems. A bi-rotator graph is obtained by making each edge of a rotator graph bi-directional. In a bi-rotator graph, average distance is improved against a rotator graph with the same number of nodes. In this paper, we give an algorithm for the container problem in bi-rotator graphs with its evaluation results. The solution achieves some fault tolerance such as file distribution based information dispersal technique. The algorithm is of polynomial order of n for an n-bi-rotator graph. It is based on recursion and divided into two cases according to the position of the destination node. The time complexity of the algorithm and the maximum length of paths obtained are estimated to be O(n3) and 4n-5, respectively. Average performance of the algorithm is also evaluated by computer experiments.

For multiple-channel active noise control (ANC) systems, distributed systems consisting of more than one controller are useful. In this paper, we propose a performance improvement algorithm for the distributed multiple-channel ANC system based on the simultaneous equations method. In the proposed algorithm, no estimation of error paths is required. This algorithm can provide good performance in canceling primary noises with auto-/cross-correlations and achieve stable noise reduction under a change of the error paths.

A millimeter wave BPF constructed from the WG mode dielectric disk resonators is presented. The design chart for the high Q WG mode resonator is obtained from Qu calculation of some WG modes. By using the design chart, high Q WG mode resonator having no influence of unwanted higher order resonances is designed. Designed resonators have different diameter and various Resonance Frequency Separation respectively. A 3 stage maximally flat BPF is constructed so that each resonator may be coupled laterally on the edge of the disk. Designed center frequency is 62.47 GHz and 3 dB bandwidth is 100 MHz. As a result, this BPF has insertion loss of 1.5 dB and some spurious responses which were existed conventional WG mode BPF are reduced considerably.

The present paper introduces a new construction of a class of binary sequence set having a zero-correlation zone (hereafter binary zcz sequence set). The cross-correlation function and the side-lobe of the auto-correlation function of the proposed sequence set is zero for the phase shifts within the zero-correlation zone. The present paper shows that such a construction generates a binary zcz sequence set by using a primitive linear recursion over GF(2), the finite field of integers modulo 2.

In this paper, a low-voltage low-power sigma-delta modulator dedicated to implantable sensing devices is presented. This second-order single-loop sigma-delta modulator is implemented with half-delay integrators. These integrators are based on new fully-differential CMOS class AB switched-Operational Transconductance Amplifier (switched-OTA). An on-chip voltage doubler is introduced to locally boost a supply voltage at the input stage of a conventional OTA in order to allow rail-to-rail signal swing. Experimental results of the modulator fabricated in CMOS 0.18 µm technology confirm its expected features of a peak signal-to-noise ratio (SNR) of 72 dB, a signal-to-noise distortion ratio (SNDR) of 62 dB in a 5 kHz signal bandwidth, and a power consumption lower than 66 µW with a 900 mV voltage supply.