In this study we propose a manufacturing control framework based on radio-frequency identification (RFID) technology and a distributed information system to construct a mass-customization production process in a loosely coupled shop-floor control environment. On the basis of this framework, we developed RFID middleware and an integrated information system for tracking and controlling the manufacturing process flow. A bicycle manufacturer was used to demonstrate the prototype system. The findings of this study were that the proposed framework can improve the visibility and traceability of the manufacturing process as well as enhance process quality control and real-time production pedigree access. Using this framework, an enterprise can easily integrate an RFID-based system into its manufacturing environment to facilitate mass customization and a just-in-time production model.

Current sub-pixel motion estimation algorithm is time and memory-consuming when performing image compression and communication. So we propose a selective interpolation method for sub-pixel motion estimation. We applied selective interpolations after estimating a candidate for sub-pixel accuracy motion vector from the simplest mathematical model. According to simulation results, the proposed method attains nearly the same performance as the full-search for half-pixel motion estimation with much lower computational complexity.

A new online, unsupervised voice activity detection (VAD) method is proposed. The method is based on a feature derived from high-order statistics (HOS), enhanced by a second metric based on normalized autocorrelation peaks to improve its robustness to non-Gaussian noises. This feature is also oriented for discriminating between close-talk and far-field speech, thus providing a VAD method in the context of human-to-human interaction independent of the energy level. The classification is done by an online variation of the Expectation-Maximization (EM) algorithm, to track and adapt to noise variations in the speech signal. Performance of the proposed method is evaluated on an in-house data and on CENSREC-1-C, a publicly available database used for VAD in the context of automatic speech recognition (ASR). On both test sets, the proposed method outperforms a simple energy-based algorithm and is shown to be more robust against the change in speech sparsity, SNR variability and the noise type.

This letter analyzes the outage probability of limited feedback beamforming systems with receive antenna selection. Tight analytical closed-form expressions of outage performance are derived for both cases, with and without spatial fading correlation, which allow for evaluation of the performance as a function of the codebook size, the level of fading correlation, and the number of transmit and receive antennas. Simulation results are also provided to verify the analysis.

Time-resolved microscopic optical second harmonic generation (TRM-SHG) imaging measurement revealed quantitatively the potential drop at the electrode contact of pentacene field effect transistors (FET). An activation of the SH signal at the edge of Ag-source electrode indicates the presence of large potential drop at pentacene-Ag contact during device operation, whereas negligible potential drop was observed at pentacene-Au contact. These findings agree with the injection characteristics of electrodes owing to the relationship between the work function of the metal and the HOMO level of pentacene.

With the development of digital TV system, how to display the NTSC signal in digital TV system is a problem. De-interlacing is an algorithm to solve it. In previous papers, using motion compensation (MC) method for de-interlacing needs lots of computation complexity and it is not easy to implement in hardware. In this paper, a content adaptive de-interlacing algorithm is proposed. Our algorithm is based on the motion adaptive (MA) method which combines the advantages of intra-field and inter-field method. We propose a block type decision mechanism to predict the video content instead of a blind processing with MC method throughout the entire frame. Additionally, in intra-field method, we propose the edge-base adaptive weight average (EAWA) method to achieve a better performance and smooth the edge and stripe. In order to demonstrate our algorithm, we implement the de-interlacing system on the DSP platform with thorough complexity analysis. Compared to MC method, we not only achieve higher video quality in objective and subjective view, but also consume lower computation power. From the profiling on CPU run-time analysis, the proposed algorithm is only one-fifth of MC method. At the DSP demonstration board, the saving ratio is about 54% to 96%.

Because of the fact that mobile communication channel changes by time, it is necessary to employ adaptive channel equalizers in order to combat the distorting effects of the channel. Least Mean Squares (LMS) algorithm is one of the most popular channel equalization algorithms and is preferred over other algorithms such as the Recursive Least Squares (RLS) and Maximum Likelihood Sequence Estimation (MLSE) when simplicity is the dominant decision factor. However, LMS algorithm suffers from poor performance and convergence speed within the training period specified by most of the standards. The aim of this study is to improve the convergence speed and performance of the LMS algorithm by adjusting the step size using fuzzy logic. The proposed method is compared with the Channel Matched Filter-Decision Feedback Equalizer (CMF-DFE) [1] which provides multi path propagation diversity by collecting the energy in the channel, Minimum Mean Square Error-Decision Feedback Equalizer (MMSE-DFE) [2] which is one of the most successful equalizers for the data packet transmission, normalized LMS-DFE (N-LMS-DFE) [3] , variable step size (VSS) LMS-DFE [4] , fuzzy LMS-DFE [5],[6] and RLS-DFE [7] . The obtained simulation results using HIPERLAN/1 standards have demonstrated that the proposed LMS-DFE algorithm based on fuzzy logic has considerably better performance than others.

In this paper, based on interleaved technique, we present a new method of constructing zero correlation zone (ZCZ) sequence sets. For any perfect sequence of length m(2k+1) with m > 2, k ≥ 0 and an arbitrary Hadamard matrix of order T > 2, the proposed construction can generate new optimal ZCZ sequence sets in which all the sequences are cyclically distinct.

In UMTS (universal mobile telecommunications system) networks upgraded with HSPA (high speed packet access) technology, the high access bandwidth and advanced mobile devices make it applicable to share large files among mobile users by peer-to-peer applications. To receive files quickly is essential for mobile users in file sharing applications, mainly because they are subject to unstable signal strength and battery failures. While many researches present peer-to-peer file sharing architectures in mobile environments, few works focus on decreasing the time spent in disseminating files among users. In this paper, we present an efficient peer-to-peer file sharing design for HSPA networks called AFAM -- Adaptive efficient File shAring for uMts networks. AFAM can decrease the dissemination time by efficiently utilizing the upload-bandwidth of mobile nodes. It uses an adaptive rearrangement of a node's concurrent uploads, which causes the count of the node's concurrent uploads to lower while ensuring that the node's upload-bandwidth can be efficiently utilized. AFAM also uses URF -- Upload Rarest First policy for the block selection and receiver selection, which achieves real rarest-first for the spread of blocks and effectively avoids the "last-block" problem in file sharing applications. Our simulations show that, AFAM achieves much less dissemination time than other protocols including BulletPrime and a direct implementation of BitTorrent for mobile environments.

A novel algorithm for source location by utilizing the time-difference-of-arrival (TDOA) of a signal received at spatially separated sensors is proposed. The algorithm is based on the constrained total least-squares (CTLS) technique and gives an explicit solution. Simulation results demonstrate that the proposed algorithm has high location accuracy and its performance is close to the Cramer-Rao lower bound (CRLB).

An adaptive per-survivor processing maximum likelihood sequence estimation (PSP-MLSE) using state-space based recursive least-squares (RLS) is proposed for rapidly time varying multi-path fading channels. Unlike PSP-MLSE using Kalman filtering, it does not require the knowledge of model statistics, and with an aid of state-space modeling, it has a robust performance to the fade rate, compared to PSP-MLSE using conventional RLS.

Proofs written in algebraic specification languages are called proof scores. The proof score approach to design verification is attractive because it provides a flexible way to prove that designs for systems satisfy properties. Thus far, however, the approach has focused on safety properties. In this paper, we describe a way to verify that designs for systems satisfy liveness properties with the approach. A mutual exclusion protocol using a queue is used as an example. We describe the design verification and explain how it is verified that the protocol satisfies the lockout freedom property.

Improving the overall performance of reliable speech communication in ultrashort wave radios over very noisy channels is of great importance and practical use. An iterative joint source-channel (de-)coding and (de-)modulation (JSCCM) algorithm is proposed for ITU-T Rec.G.729EV by both exploiting the residual redundancy and passing soft information throughout the receiver while introducing a systematic global iteration process. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce additional bandwidth expansion and transmission delay. Simulations show substantial error correcting performance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the JSCCM algorithm.

Recently much attention has been paid to quantum circuit design to prepare for the future "quantum computation era." Like the conventional logic synthesis, it should be important to verify and analyze the functionalities of generated quantum circuits. For that purpose, we propose an efficient verification method for quantum circuits under a practical restriction. Thanks to the restriction, we can introduce an efficient verification scheme based on decision diagrams called Decision Diagrams for Matrix Functions (DDMFs). Then, we show analytically the advantages of our approach based on DDMFs over the previous verification techniques. In order to introduce DDMFs, we also introduce new concepts, quantum functions and matrix functions, which may also be interesting and useful on their own for designing quantum circuits.

In approximately synchronous CDMA (AS-CDMA) systems, zero correlation zone (ZCZ) sequences are known as the sequences to eliminate co-channel and multi-path interferences. Therefore, numerous constructions of zero correlation zone (ZCZ) sequences have been introduced e.g. based on perfect sequences and complete complementary codes etc. However, the previous construction method which based on complete complementary code is lacking for merit figures when none of whose elements are zero. In this paper, a new construction method of ZCZ sequences based on complete complementary codes is proposed. By proposed method, non zero elements ZCZ sequences whose merit figure is greater than 1/2 are constructable.

This work proposes new features to improve the pathological voice quality classification performance. They are the means, the variances, and the perturbations of the higher-order statistics (HOS) such as the skewness and the kurtosis. The HOS-based features show meaningful differences among normal, grade 1, grade 2, and grade 3 voices classified in the GRBAS scale. The jitter, the shimmer, the harmonic-to-noise ratio (HNR), and the variance of the short-time energy are utilized as the conventional features. The performances are measured by the classification and regression tree (CART) method. Specifically, the CART-based method by utilizing both the conventional features and the HOS-based ones shows its effectiveness in the pathological voice quality measurement, with the classification accuracy of 87.8%.

In this paper we introduce new M-ary sequences of length pq, called generalized M-ary related-prime sequences, where p and q are distinct odd primes, and M is a common divisor of p-1 and q-1. We show that their out-of-phase autocorrelation values are upper bounded by the maximum between q-p+1 and 5. We also construct a family of generalized M-ary related-prime sequences and show that the maximum correlation of the proposed sequence family is upper bounded by p+q-1.

The weakness of implementation for LDPC encoder is that conventional binary Matrix Vector Multiplier has many clock cycles which lead to limited throughput. In this letter in order to construct efficient architecture, we target on IEEE 802.16e LDPC encoders. Over the standard H matrices with Circulant Permutation Matrices, we propose semi-parallel architecture by using cyclic right shift registers and exclusive-OR instead of complex Matrix Vector Multipliers. Proposed efficient encoder for IEEE 802.16e LDPC satisfies compact size and high throughput.

In the context of multiple constant multiplication (MCM) design, we propose a novel common sub-expression elimination (CSE) algorithm that models the optimal synthesis of coefficients into a 0-1 mixed-integer linear programming (MILP) problem with a user-defined generic logic depth constraint. We also propose an efficient solution space, which combines all minimal signed digit (MSD) representations and the shifted sum (difference) of coefficients. In the examples we demonstrate, the combination of the proposed algorithm and solution space gives a better solution comparing to existing algorithms.

In multi-FPGA prototyping systems for circuit verification, serialized time-multiplexed I/O technique is used because of the limited number of I/O pins of an FPGA. The verification time depends on a selection of inter-FPGA signals to be time-multiplexed. In this paper, we propose a method that minimizes the verification time of multi-FPGA systems by finding an optimal selection of inter-FPGA signals to be time-multiplexed. In the experiments, it is shown that the estimated verification time is improved 38.2% on average compared with conventional methods.