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.

We proposed a fuzzy control scheme to implement the cycle-to-cycle control for restoring swing phase of gait using functional electrical stimulation (FES). We designed two fuzzy controllers for the biceps femoris short head (BFS) and the vastus muscles to control flexion and extension of the knee joint during the swing phase. Control capabilities of the designed fuzzy controllers were tested and compared to proportional-integral-derivative (PID) and adaptive PID controllers in automatic generation of stimulation burst duration and compensation of muscle fatigue through computer simulations using a musculo-skeletal model. Parameter adaptations in the adaptive PID controllers did not significantly improve the control performance of the PID controllers. The fuzzy controllers were superior to the PID and adaptive PID controllers under several subject conditions and different fatigue levels. These results showed the fuzzy controller would be suitable to implement the cycle-to-cycle control of FES-induced gait.

In this paper, we propose an OFDM scheme with pre-IDFT/DFT and the frequency domain equalization on frequency-selective Rayleigh fading channels. In this scheme, a two-dimensional block interleaving is used to randomize the correlated noise caused by the frequency domain linear equalizer. Then, the pre-DFT averages the interleaved noise enhancement and improves the error performance of the proposed scheme. Computer simulations confirm the bit error probability of the proposed scheme for multilevel modulations.

A mutated adder architecture utilizing a mixture of carry propagation schemes is proposed to design a delay-area efficient adder which were not available in an ordinary design space. Further, we develop an optimization method based on integer linear programming to search the expanded design space of the mutated adder.

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.

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.

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.

In an empowered Internet with end-to-end Quality of Service (QoS) facilities, it is essential for content servers to minimize reserved network resources to achieve a reduction in transmission cost for the use of QoS. Resource reservation usage charging forces customers to efficiently use network resources. In this paper, we analyze a model that optimizes the total cost when semi-elastic traffic flows are delivered in a client-server scenario. The client uses the proposed analytical model to easily control its buffer occupancy and to determine at any time if it is needed to utilize resource reservation or best-effort transmission mode.

Severe frequency-selective fading, encountered in a broadband wireless mobile communication, significantly degrades the bit error rate (BER) performance of direct sequence spread spectrum (DSSS) signal transmission with rake combining. In this paper, frequency-domain pre-equalization transmission, called pre-FDE transmission, is presented for orthogonal multicode DSSS signal transmission. It is confirmed by the computer simulation that pre-FDE transmission can achieve a BER performance almost identical to that attainable by FDE reception.

In order to reduce the size of a wireless system, we propose a design theory for the broadband impedance matching circuit which connects an electrically small antenna (ESA) to a semiconductor amplifier. We confirmed its validity for the case of connection between a small slot loop antenna with a small radiation resistance of Ra =0.776 Ω and a semiconductor amplifier with high input impedance of ZL =321-j871 Ω with the aid of the simulations by the electrical circuits using transmission lines as well as the electromagnetic field (EM field) simulator. We also made experiments on this antenna with matching circuits using high temperature superconductor YBCO thin films on MgO substrates.

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.

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.

The latest video coding standard, H.264/AVC, adopts 44 approximate transform instead of 88 discrete cosine transform (DCT) to avoid the inverse transform mismatch problem. However, that is only one of the factors that make it difficult to transcode pre-coded video contents with the previous standards to H.264/AVC in the common domain without causing cascaded pixel-domain transcoding. In this paper, to support the existent DCT-domain transcoding schemes and to reduce computational complexity, we propose an efficient algorithm that converts the quantized 88 DCT block into four newly quantized 44 transformed blocks. The experimental results show that the proposed scheme reduces computational complexity by 5-11% and improves video quality by 0.1-0.5 dB compared with the cascaded pixel-domain transcoding scheme that exploits inverse quantization (IQ), inverse DCT (IDCT), DCT, and re-quantization (re-Q).

This paper presents a novel method of designing microstrip line multi-frequencies band filters by applying the SIR (stepped impedance resonators) technology. Utilizing the S-parameter and the ABCD parameters of a two-port network is for the analysis of short-circuited and open-circuited resonators with various combinations of series and shunt sequences. By controlling the impedance ratio of the resonators, both center frequencies of the two passbands then are determined. Moreover, a global synthesis approach is also discussed on miniaturization. A simplified architecture based on bent SIR offers the 50% area reduction of layout. Technology of matching circuit creates higher performance multi-band filter. We adjust impedance and electrical length of transmission line (TL) to compensate multi-band and bending for matches and highly improve the insertion and reflection loss. Simulation and measurement are performed to validate our method and are pretty matched.

In this paper, two techniques are proposed for accelerating and stabilizing the Levenberg-Marquardt (LM) method where its conventional stabilizer matrix (identity matrix) is superseded by (1) a diagonal matrix whose elements are column norms of Jacobian matrix J, or (2) a non-diagonal square root matrix of J TJ. Geometrically, these techniques make constraint conditions of the LM method fitted better to relevant cost function than conventional one. Results of numerical simulations show that proposed techniques are effective when both column norm ratio of J and mutual interactions between arguments of the cost function are large. Especially, the technique (2) introduces a new LM method of damped Gauss-Newton (GN) type which satisfies both properties of global convergence and quadratic convergence by controlling Marquardt factor and can stabilize convergence numerically. Performance of the LMM techniques are compared also with a damped GN method with line search procedure.

This letter presents a synchronizer and its handshake interface for bridging clock domains in SoC. The proposed scheme uses a double two-flop synchronizer operated at different clock edges respectively, based on a two-phase handshake protocol. Performance analysis shows that the proposed design reduces latency up to a clock cycle, while retaining its safety to a tolerable level.

In this letter, we consider a problem of global exponential stabilization of a class of approximately feedback linearized systems. With a newly proposed LMI-condition, we propose a controller design method which is shown to be improved over the existing methods in several aspects.

The actual sound environment system exhibits various types of linear and non-linear characteristics, and it often contains an unknown structure. Furthermore, the observations in the sound environment are often in the level-quantized form. In this paper, a method for estimating the specific signal for stochastic systems with unknown structure and the quantized observation is proposed by introducing a system model of the conditional probability type. The effectiveness of the proposed theoretical method is confirmed by applying it to the actual problem of psychological evaluation for the sound environment.

This letter discusses the prediction of the time-varying bit error rate (BER) for a transmitting channel using recent transmissions and retransmissions. Depending on the predicted BER, we propose a maximum frame size control to improve the goodput in wireless networks. It is shown, using simulation, that when the maximum frame size is controlled relative to the time-varying BER the goodput of the network is improved.

We propose a new call admission control (CAC) scheme for voice calls in cellular mobile communication networks. It is assumed that the rejection of a hand-off call is less desirable than that of a new call, for a hand-off call loss would cause a severe mental pain to a user. We consider the pains of rejecting new and hand-off calls as different costs. The key idea of our CAC is to restrict the admission of new calls in order to minimize the total expected costs per unit time over the long term. An optimal policy is derived from a semi-Markov decision process in which the intervals between successive decision epochs are exponentially distributed. Based on this optimal policy, we calculate the steady state probability for the number of established voice connections in a cell. We then evaluate the probability of blocking new calls and the probability of forced termination of hand-off calls. In the numerical experiments, it is found that the forced termination probability of hand-off calls is reduced significantly by our CAC scheme at the slight expense of the blocking probability of new calls and the channel utilization. Comparison with the static guard channel scheme is made.