We developed two types of practical maximum-likelihood detectors (MLD) for multiple-input multiple-output (MIMO) systems, using a field programmable gate array (FPGA) device. For implementations, we introduced two simplified metrics called a Manhattan metric and a correlation metric. Using the Manhattan metric, the detector needs no multiplication operations, at the cost of a slight performance degradation within 1 dB. Using the correlation metric, the MIMO-MLD can significantly reduce the complexity in both multiplications and additions without any performance degradation. This paper demonstrates the bit-error-rate performance of these MLD prototypes at a 1 Gbps-order real-time processing speed, through the use of an all-digital baseband 44 MIMO testbed integrated on the same FPGA chip.

Space-time coding (STC) schemes for communication systems employing multiple transmit and receive antennas have received considerable interest recently. On space-time coding, some algorithms with perfect channel state information (CSI) have been proposed. In certain fast varying situation, however, it may be difficult to estimate the channel accurately and it is natural to study the blind detection algorithm without CSI. In this paper, based on subspace, a new blind detection algorithm without CSI is proposed. Using singular value decomposition (SVD) on output signal, noise subspace and signal subspace, which keep orthogonal to each other, are obtained. By searching the intersection of the signal subspace and the limited symbol vector set, symbol detection is achieved. The simulations illustrate that the proposed algorithm significantly improves system performance by receiving more output signals relative to transmit symbols. Furthermore, the presented algorithm is robust to the fading channel that changes between two successive blocks.

Proportional fair bandwidth allocation in packet switches is a fundamental issue to provide quality of service (QoS) support in IP networks. In input-queued switches, packet-mode scheduling delivers all the segments of a packet contiguously from the input port to the output port, thus greatly simplifying the design of packet reassembly modules and yielding performance advantage over cell-mode scheduling under certain conditions [1]. One of the important issues of packet-mode scheduling is how to achieve fair bandwidth allocation among flows with different packet sizes. This paper presents an algorithm called packet-mode fair scheduling (pFS) that guarantees each flow a bandwidth proportional to its reservation regardless of the packet size distribution and the system load. Simulations show that our approach achieves good fairness as well as high throughput and low packet delay. Compared to algorithms without fairness mechanism, pFS yields significant performance improvement in terms of average packet delay when the traffic is heterogeneous. A hardware implementation is presented to show that the proposed algorithm has low complexity and the computation can be completed in a single clock cycle, which makes pFS applicable to high-speed switches.

A novel digital redesign methodology based on evolutionary programming (EP) is introduced to find the 'best' digital controller for optimal tracking design of hybrid uncertain multi-input/ multi-output (MIMO) input-delay systems with constraints on states and controls. To deal with these multivariable concurrent specifications and system restrictions, instead of conventional interval methods, the proposed global optimization scheme is able to practically implement optimal digital controller for constrained uncertain hybrid systems with input time delay. Further, an illustrative example is included to demonstrate the efficiency of the proposed method.

Currently, the automatic image inpainting methods emphasize the inpainting techniques either globally or locally. They didn't consider the merits of global and local techniques to compensate each other. On the contrary, the artists fixed an image in global view firstly, and then focus on the local features of it, when they repaired it. This paper proposes a progressive processing of image inpainting method based on multi-resolution analysis. In damaged and defective area, we imitate the artistic techniques to approach the effectiveness of image inpainting in human vision. First, we use the multi-resolution characteristics of wavelet transform, from the lowest spatial-frequency layer to the higher one, to analyze the image from global-area to local-area progressively. Then, we utilize the variance of the energy of wavelet coefficients within each image block, to decide the priority of inpainting blocks. Finally, we extract the multi-resolution features of each block. We take account of the correlation among horizontal, vertical and diagonal directions, to determine the inpainting strategy for filling image pixels and approximate a high-quality image inpainting to human vision. In our experiments, the performance of the proposed method is superior to the existing methods.

This letter proposes two very-low-complexity maximum-likelihood (ML) detection algorithms based on QR decomposition for the quasi-orthogonal space-time code (QSTBC) with four transmit antennas [3]-[5], called VLCMLDec1 and VLCMLDec2 decoders. The first decoder, VLCMLDec1, can be used to detect transmitted symbols being extracted from finite-size constellations such as phase-shift keying (PSK) or quadrature amplitude modulation (QAM). The second decoder, VLCMLDec2, is an enhanced version of the VLCMLDec1, developed mainly for QAM constellations. Simulation results show that both of the proposed decoders enable the QSTBC to achieve ML performance with significant reduction in computational load.

This paper presents the novel method not only to suppress the input current harmonics but also to realize the low frequency output voltage ripple using the multiple-input ac-dc converter, which is considered from viewpoints of the relatively small power application and simple circuit configuration. The operation principle and control strategy of the proposed circuit are discussed. As a result, it is clarified that the new circuit has excellent performance characteristics such as high power factor over 0.99, low total harmonic current distortion factor less than 9.2% and low output voltage ripple of 40 mV.

Orthogonal space-time block codes (STBCs) appear to be a very fascinating means of enhancing reception quality in quasi-static MIMO channels due to their full diversity, and especially their simple maximum-likelihood (ML) decoders. However, full-rate full-diversity orthogonal STBCs do not exist for more than two transmit antennas. Vertical layered space-time architecture (so-called the V-BLAST) with a nulling- and cancelling-based detection algorithm, in contrast, has an ability of achieving high transmission rates at the cost of having very low diversity gain, an undesirable consequence caused by the interference nulling and cancelling processes. The uncoded V-BLAST system is able to reach its ML performance with the aid of the sphere decoder algorithm at the expense of higher detection complexity. Undoubtedly, the tradeoff between transmission rates, diversity, and complexity is inherent in designing space-time codes. This paper investigates a method to increase the "nulling diversity gains" for a general high-rate space-time code and introduces a new design strategy for high-rate space-time codes detected based on interference nulling and cancelling processes, thanks to which high-rate quasi-orthogonal space-time codes for MIMO applications are proposed. We show that when nT transmit and nR=nT receive antennas are deployed, the first code offers a transmission rate of (nT-1) with a minimum nulling diversity order of 3, whereas the second one offers a transmission rate of (nT-2) with a minimum nulling diversity order of 5. Therefore, the proposed codes significantly outperform the V-BLAST as nR=nT. Simulation results and discussions on the performance of the proposed codes are provided.

In order to control a sound field using multiple sources and microphones, we must choose the optimum values of parameters such as the numbers of sources and microphones, the location of the sources and the microphones and the filter tap length. Because there is a huge number of possible combinations of these conditions, the boundary surface control principle can be useful as a basis of a design method of such a system. In this paper, a design method of sound field reproduction and active noise control based on the BSC principle are described and several example of its application are presented.

A method to monitor the GaInP/GaAs HBT device structure including emitter ledge thickness is demonstrated in this paper. The base thickness and base doping density are obtained through base transit time and base sheet resistance measurements while the base transit time is measured through the cut-off frequency measurements at various bias points. A large size two-emitter HBT device is used to measure the ledge thickness. Emitter doping profile and collector doping profile are obtained by the large size HBT device through C-V measurements. An FATFET device formed by two emitters as drain and source terminals and the interconnect metal as the on-ledge Schottky gate between two emitters is used to measure the ledge thickness.

The discovery of the Turbo codes has driven research on the creation of new signal detection concepts that are, in general, referred to as the Turbo approach. Recently, this approach has made a drastic change in creating signal detection techniques and algorithms such as equalization of inter-symbol interference (ISI) experienced by broadband single carrier signaling over mobile radio channels. A goal of this paper is to provide readers with broad views and knowledge of the Turbo concept-based Multiple-Input Multiple-Output (MIMO) signal transmission techniques. How the techniques have been developed in various applications and how they perform in real-field environments are introduced.

A 3-bit flash analog-to-digital converter (ADC) for electronic dispersion compensation (EDC) was developed using InP HBTs. Nyquist operation was confirmed up to 24 Gsps, which enables oversampling acquisition for 10 Gbit/s non-return-to-zero (NRZ) signals. The ADC can also be operated at up to 37 Gsps for low input frequencies. To reduce aperture jitter and achieve a wide band of over 7 GHz, an analog input signal for all pre-amplifiers and a clock signal for all latched comparators are provided as traveling waves through coplanar transmission lines. EDC was demonstrated by capturing a 10-Gbit/s pseudo-random bit stream (PRBS) with the waveform degraded by polarization-mode dispersion (PMD). By using the captured data, we confirmed that a calculation of a transversal filter mitigates PMD.

A simple high-speed low input impedance CMOS current comparator is presented. The circuit uses improved Wilson current-mirror to perform subtraction. Negative feedback is employed to reduce the input impedance of the circuit. HSPICE is used to verify the circuit performance with standard 0.5 µm CMOS technology. Simulation results demonstrate propagation delay of 1.02 ns, average power consumption of 0.9 mW, and input impedance of 137 Ω for 0.1 µA input current at the supply voltage of 3 V.

A GaInAs/InP multiple quantum well (MQW)-based wavelength demultiplexer composed of a waveguide array in which the refractive index varies across the array yielded successful results of wavelength demultiplexing and optical deflection. Since optical path length differences between waveguides in the array are achieved through refractive-index differences controlled by the SiO2 mask design in selective metal-organic vapor phase epitaxy (MOVPE), a straight waveguide grating can be formed with reduced optical propagation losses. A straight waveguide array device with a 1.4% refractive-index difference was fabricated. The fabrication of a preliminary wavelength demultiplexer was also achieved, for which a wavelength separation with an approximately 25 nm spacing and free spectral range (FSR) of approximately 100 nm were obtained. Moreover, an optical deflector was investigated and primitive deflection was achieved at 1460 and 1490 nm incident wavelengths.

A new interacting multiple model (IMM) algorithm using intelligent input estimation (IIE) is proposed for maneuvering target tracking. In the proposed method, the acceleration level for each sub-model is determined by IIE-the estimation of the unknown target acceleration by a fuzzy system using the relation between the residuals of the maneuvering filter and the non-maneuvering filter. The genetic algorithm (GA) is utilized to optimize a fuzzy system for a sub-model within a fixed range of target acceleration. Then, multiple models are represented as the acceleration levels estimated by these fuzzy systems, which are optimized for different ranges of target acceleration. In computer simulation for an incoming anti-ship missile, it is shown that the proposed method has better tracking performance compared with the adaptive interacting multiple model (AIMM) algorithm.

We have developed a traveling-wave optical modulator using an n-i-n heterostructure fabricated on an InP substrate. The modulation characteristics are studied theoretically and experimentally. We obtained an extremely small π voltage (Vπ) of 2.2 V, even for a short signal-electrode length of 3 mm. We confirmed a wide frequency bandwidth and clearly open eye diagrams at 40 Gbit/s.

A hybrid beamformer composed of a direction-of-arrival (DOA) based scheme and maximal ratio combining (MRC) is proposed to overcome the degradation caused by inaccurate channel estimation due to insufficient pilot power, which happens in conventional single-input, multiple-output (SIMO) Code Division Multiple Access (CDMA) reverse link. The proposed scheme can provide more accurate channel estimation and interference reduction at the expense of diversity gain in the spatially correlated SIMO channel. As a result, the hybrid scheme outperforms conventional MRC beamformers for six or more antennas in the channel environment, in which the angle-of-spread (AOS) is within 30.

In the design of cascode CMOS low-noise amplifiers, the gate-drain capacitance is generally neglected because it is thought to be small enough compared to gate-source capacitance. However, a careful examination will reveal the fact that the drain impedance of the input transistor significantly affects the input impedance through the gate-drain capacitance, especially as the CMOS technology getting more and more advanced. Moreover, the substrate coupling network of the input transistor also comes into play when the drain impedance of the input transistor is high enough compared to the substrate coupling network. In order to make input matching easier, it is desirable to know the details of the substrate coupling network. Unfortunately, designers generally do not have enough information about the technology they have used, not to mention knowing the details concerning the substrate coupling network. As a matter of fact, designers generally do have foundry provided component models that contain information about the substrate coupling network. This gives us the chance to minimize its effect and predict the input impedance of a low noise amplifier more accurately. In this paper, we show that the effect of the substrate coupling network can be ignored by keeping the drain impedance of the input transistor low enough and a proper drain impedance can then be chosen to achieve input matching without the need of iteration steps. Simulation results of a 2.4 GHz CMOS low noise amplifier using foundry provided component models are also presented to demonstrate the validation of the proposed input matching method.

Current literature on input buffer management reveals that, in representative ATM networks under highly bursty traffic conditions, the fuzzy thresholding approach yields lower cell loss rate at the cost of lower throughput. Also, under less bursty traffic, the traditional fixed thresholding approach achieves higher throughput at the expense of higher cell loss rate. The integration of these two properties into practice is termed adaptive dynamic buffer management (ADBM) approach for input buffers and its assessment is the objective of this paper. The argument is that, given that the traffic conditions are constantly changing, to achieve efficiency during actual operation, the network control must dynamically switch, at every ATM switch, under the call processor's control, between the two input buffer management techniques, dictated by the nature of the traffic at the inputs of the corresponding switch. The need to involve the call processor marks the first effort in the literature to dynamically configure input buffer management architectures at the switch fabric level under higher level call processor control. It stems from the fact that the switch fabric operates very fast and cannot engage in complex decision making without incurring stiff penalty. To achieve this goal, the network control needs knowledge of the burstiness of the traffic at the inputs of every ATM switch. The difficulties with this need are two-fold. First, it is not always easy to obtain the traffic model and model parameters for a specific user's call. Second, even where the traffic model and the model parameters are known for a specific user's call, this knowledge is valid only at the source switch where the user interfaces with the network. At all other switches in the network, the cells of the traffic in question interact asynchronously with the cells from other traffic sources and are subject to statistical multiplexing. Thus, to obtain the exact nature of the composite traffic at the inputs of any ATM switch, is a challenge. Conceivably, one may determine the burstiness by counting the number of cells incurred at the inputs of an ATM switch over a defined time interval. The challenge posed by this proposition lies in the very definition of burstiness in that the time interval must approach, in the limit, zero or the resolution of time in the network. To address this challenge, first, a 15-node representative ATM network is modeled in an asynchronous, distributed simulator and, second, simulated on a network of workstations under realistic traffic stimuli. Third, burstiness indices are measured for the synthetic, stochastic traffic at the inputs of every ATM switch as a function of the progress of simulation for different choices of time interval values, ranging from 20,000 timesteps down to 1,000 timesteps. A timestep equals 2.73 µs. Results reveal that consistent burstiness indices are obtained for interval choices between 1,000 and 5,000 timesteps and that a burstiness index of 25, measured at 3,000 timestep interval, constitutes a reasonable and practical threshold value that distinguishes highly bursty traffic that warrants the use of the fuzzy thresholding approach from less bursty traffic that can benefit from the fixed thresholding scheme. A comparative performance analysis of ADBM yields the following. For pure fixed and pure fuzzy thresholding schemes, the throughputs are at 73.88% and 71.53% while the cell drop rates are at 4.31% and 2.44%,respectively. For the ADBM approach, where the input buffer management alternates at each individual ATM switch between the fixed and fuzzy schemes, governed by measured burstiness index threshold of 25 for a 3,000 timestep interval, the throughput is 74.77%, which is higher than even the pure fixed scheme while the cell drop rate is 2.21% that is lower than that of the pure fuzzy scheme. In essence, ADBM successfully integrates the best characteristics of the fuzzy and fixed thresholding schemes.

This paper presents a novel decoding algorithm for turbo codes, in which the likelihood values for redundant parts are updated in order for those values to become more reliable. A criterion for updating the redundant likelihood values is proposed, which is based on the comparisons of the channel values with the re-generated values by the soft-input and soft-output encoders. It is shown that the proposed method can improve the error correcting capabilities, i.e., the improvement of BER/BLER performance and the achievable BER limit.