In this paper, we use a modified Gaussian filter to improve enlargement accuracy of the arbitrary scale LP enlargement method, which is based on the Laplacian pyramid representation (so called "LP method"). The parameters of the proposed algorithm are extracted through a theoretical analysis and an experimental estimation. Experimental results show that the proposed modified Gaussian filter is effective for the arbitrary scale LP enlargement method.

A nonlinear harmonic estimator (NHE) is proposed for extracting a harmonic signal and its fundamental frequency in the presence of white noise. This estimator is derived by applying an extended complex Kalman filter (ECKF) to a multiple sinusoidal model with state-representation and then efficiently specializing it for the case of harmonic estimation. The effectiveness of the NHE is verified using computer simulations.

The physical origin of stress-induced leakage currents (SILC) in ultra-thin SiO2 films is described. Assuming a two-step trap-assisted tunneling process accompanied with an energy relaxation process of trapped electrons, conditions of trap sites which are origin of SICL are quantitatively found. It is proposed that the trap site location and the trap state energy can be explained by a mean-free-path of hole in SiO2 films and an atomic structure of the trap site by the O vacancy model.

To overcome the demerits of two passive optical networks; the small link capacity of the TDM-PON, and the ineffective link utilization of the WDM-PON; we propose a novel access network architecture featuring a WDM-based feeder network and a TDM-based distribution network. In this paper, we examine the design issues of the key constituent of SWE-PON (Scalable WDM-based Ethernet hybrid-PON) to validate its economic and practical feasibility. For flawless network operation, the wavelength tuning rule is investigated so that it does not collide between wavelengths from the tunable lasers belonging to the WDM coupler. Also, the potential problem between the tunable laser and the reflective operational device is analyzed in detail. From the numerical analysis and simulation, we demonstrate the variation of the network performance in terms of the upstream traffic delay and throughput of the ONU in accordance with the sharing structure of distribution network and the number of tunable laser devices (TLDs) at the feeder network.

In this Letter, we propose a least mean square (LMS) with adaptive step-size (AS) algorithm for adaptive blind carrier frequency offset (CFO) estimation in the orthogonal frequency division multiplexing system. In conjunction with the closed-loop estimate structure, the proposed algorithm eliminates the inter-carrier interference caused by time varying CFO. To improve the convergence performance of the fixed step-size LMS estimator, the regular AS LMS algorithm offers faster convergence speed and more accuracy to the CFO estimate. Several computer simulation examples are presented for illustrating the effectiveness of the proposed algorithm.

We experimentally investigated terahertz photomixing operation at room temperature in an InGaP/InGaAs/ GaAs two-dimensional plasmon-resonant photomixer incorporating grating-bicoupled dual-gate structure. Photoelectrons drifting into a high-density plasmon cavity grating from an adjacent low-density one extensively excite the plasmon resonance, resulting in emission of terahertz radiation. A vertical cavity formed between the two-dimensional plasmon grating plane and an indium-tin-oxide mirror at the back surface gains the radiation. Self-oscillation initially at around 4.5 THz excited by a dc-photo carrier component was reinforced by the photomixed differential-frequency excitation at 4.0 and 5.0 THz. This indicates a possibility of injection-locked oscillation of the photomixer in the terahertz frequency band.

A two-dimensional (2-D) physical model of n-channel poly-Si LDD TFTs in comparison with that of SD TFTs is presented to analyze hot-carrier degradation. The model is based on 2-D device simulator's Gaussian doping profiles for the source and drain junctions fitted to the lateral and vertical impurity profiles in poly-Si obtained from a 2-D process simulator. We have shown that, in the current saturation bias (Vg

A new block coded modulation scheme with inter-level memory is proposed. The proposed code construction is based on the use of single parity check codes to concatenate a set of coded blocks. Simulation results show that the proposed scheme can achieve considerable coding gains while the decoding complexity is not too large.

To keep in step with the rapid progress of high quality imaging systems, the Digital Cinema Initiative (DCI) has been issuing digital cinema standards that cover all processes from production to distribution and display. Various evaluation measurements are used in the assessment of image quality, and, of these, the required number of quantization bits is one of the most important factors in realizing the very high quality images needed for cinema. While DCI defined 12 bits for the bit depth by applying Barten's model to just the luminance signal, actual cinema applications use color signals, so we can say that this value has an insufficient theoretical basis. This paper, first of all, investigates the required number of quantization bits by computer simulations in discrete 3-D space for the color images defined using CIE's XYZ signal. Next, the required number of quantization bits is formulated by applying Taylor's development in the continuous value region. As a result, we show that 13.04 bits, 11.38 bits, and 10.16 bits are necessary for intensity, density, and gamma-corrected signal quantization, respectively, for digital cinema applications. As these results coincide with those from calculations in the discrete value region, the proposed analysis method enables a drastic reduction in the computer simulation time needed for obtaining the required number of quantization bits for color signals.

The present paper introduces the construction of a class of sequence sets with zero-correlation zones called zero-correlation zone sequence sets. The proposed zero-correlation zone sequence set can be generated from an arbitrary perfect sequence, the length of which is longer than 4. The proposed sets of ternary sequences, which can be constructed from an arbitrary perfect sequence, can successfully provide CDMA communication without co-channel interference. In an ultrasonic synthetic aperture imaging system, the proposed sequence set can improve the signal-to-noise ratio of the acquired image.

Transmission schemes that gain content from multiple servers concurrently have been highlighted due to their ability to provide bandwidth aggregation, stability on dynamic server departure, and load balancing. Previous approaches employ parallel downloading in the transport layer to minimize the receiver buffer size and maximize bandwidth utilization. However, they only focus on the receiver operations and induce considerable overhead at the senders in contradiction to the main goal of a multi-provider environment, offloading popular servers through replication. In the present work, the authors propose MTCP, a novel transport layer protocol that focuses on reduction of the sender overhead through the elimination of unnecessary disk I/Os and efficient buffer cache utilization. MTCP also balances trade-off objectives to minimize buffering at receivers and maximize the request locality at senders.

This paper describes a novel parameter generation algorithm for an HMM-based speech synthesis technique. The conventional algorithm generates a parameter trajectory of static features that maximizes the likelihood of a given HMM for the parameter sequence consisting of the static and dynamic features under an explicit constraint between those two features. The generated trajectory is often excessively smoothed due to the statistical processing. Using the over-smoothed speech parameters usually causes muffled sounds. In order to alleviate the over-smoothing effect, we propose a generation algorithm considering not only the HMM likelihood maximized in the conventional algorithm but also a likelihood for a global variance (GV) of the generated trajectory. The latter likelihood works as a penalty for the over-smoothing, i.e., a reduction of the GV of the generated trajectory. The result of a perceptual evaluation demonstrates that the proposed algorithm causes considerably large improvements in the naturalness of synthetic speech.

Acoustic field analysis results of surface acoustic wave dispersive delay lines using inclined chirp IDTs on a Y-Z LiNbO3 substrate are described. The calculated results are compared with optical measurements. The angular spectrum of the plane wave method is applied to calculation of the acoustic fields considering the anisotropy of the SAW velocity by using the polynomial approximation. Acoustic field propagating along the Z-axis of the substrate, which is the main beam excited by the inclined chirp IDT, shows asymmetric distribution between the +Z and -Z directions. Furthermore the SAW beam propagating in a slanted direction with an angle of +18 from the Z axis to the X-axis is observed. It is described that the SAW beam propagating in a slanted direction is the first side lobe excited by the inclined chirp IDT. The acoustic field shows asymmetric distribution along the X-axis because of the asymmetric structure of the inclined chirp IDT. Finally, acoustic field of a two-IDT connected structure which consists of the same IDTs electrically connected in series is presented. The acoustic field of the two-IDT connected structure is calculated to be superposed onto the calculated result of the acoustic field exited by one IDT on the calculated result shifted along the X-axis. Two SAW beams excited by IDTs are observed. The distributions of the SAW beams are not in parallel. The calculated results show good agreement with the optical measurement results.

An adaptive array code acquisition for direct-sequence/code-division multiple access (DS/CDMA) systems was recently proposed to enhance the performance of the conventional correlator-based method. The scheme consists of an adaptive spatial and an adaptive temporal filter, and can simultaneously perform beamforming and code-delay estimation. Unfortunately, the scheme uses a least-mean-square (LMS) adaptive algorithm, and its convergence is slow. Although the recursive-least-squares (RLS) algorithm can be applied, the computational complexity will greatly increase. In this paper, we solve the dilemma with a low-complexity conjugate gradient (LCG) algorithm, which can be considered as a special case of a modified conjugate gradient (MCG) algorithm. Unlike the original conjugate gradient (CG) algorithm developed for adaptive applications, the proposed method, exploiting the special structure inherent in the input correlation matrix, requires a low computational-complexity. It can be shown that the computational complexity of the proposed method is on the same order of the LMS algorithm. However, the convergence rate is improved significantly. Simulation results show that the performance of adaptive array code acquisition with the proposed CG algorithm is comparable to that with the original CG algorithm.

A precoding scheme is described for multiple-input and multiple-output orthogonal frequency-division multiplexing systems with a QR-decomposition maximum likelihood detector (MLD) incorporated with a parallel interference canceller (PIC) at a receiver. Transmit antenna ranking based on received substream signal power or per-substream minimum Euclidean distances is fed back to a transmitter. Based on the ranking information, precoding matrices are determined as permutation matrices such that specific packets are transmitted from transmit antennas with higher channel quality over the whole subcarriers. The simulation results demonstrated that precoding effectively utilizes PIC by reducing the possibility that all substreams are incorrectly decoded and thus improves the transmission performance of a QR-decomposition MLD with PIC.

We propose and experimentally demonstrate a simple method for monitoring optical signal-to-noise ratio. The novel method can be used in the optical transport networks using optical cross-connects or reconfigurable optical add-drop multiplexers. OSNR is measured by monitoring the transmitted optical power and the reflected optical power from fiber Bragg grating. We have obtained OSNR with an error less than 0.8 dB.

One approach to accommodating IP traffic on a wavelength division multiplexing (WDM) network is to construct a logical topology, establishing a set of lightpaths between nodes. The lightpaths carry IP traffic but do not require any electronic packet processing at intermediate nodes, thereby reducing the load on those nodes. When the IP and WDM networks have independent routing functions, however, the lightpaths in the WDM network may not be fully utilized by the IP router. It is therefore necessary to integrate the two routing mechanisms in order to utilize resources efficiently and adapt to changes in traffic. In this paper, we propose an integrated routing mechanism for IP over WDM networks. The key idea is to first prepare a set of virtual-links representing the lightpaths that can be established by the WDM network, then calculate the minimum cost route on an IP network including those links. Our simulation results show that when traffic patterns do not change, the throughput of our method is almost the same as that of a logical topology optimally designed for a given traffic demand. When traffic patterns change, the throughput of our method is about 50% higher than that of the logical topology.

Two-dimensional discrete wavelet transform (DWT) for processing image is conventionally designed by line-based architectures, which are simple and have low complexity. However, they suffer from two main shortcomings - the memory required for storing intermediate data and the long latency of computing wavelet coefficients. This work presents a new block-based architecture for computing lifting-based 2-D DWT coefficients. This architecture yields a significantly lower buffer size. Additionally, the latency is reduced from N2 down to 3N as compared to the line-based architectures. The proposed architecture supports the JPEG2000 default filters and has been realized in ARM-based ALTERA EPXA10 Development Board at a frequency of 44.33 MHz.

A statistical speech synthesis system based on the hidden Markov model (HMM) was recently proposed. In this system, spectrum, excitation, and duration of speech are modeled simultaneously by context-dependent HMMs, and speech parameter vector sequences are generated from the HMMs themselves. This system defines a speech synthesis problem in a generative model framework and solves it based on the maximum likelihood (ML) criterion. However, there is an inconsistency: although state duration probability density functions (PDFs) are explicitly used in the synthesis part of the system, they have not been incorporated into its training part. This inconsistency can make the synthesized speech sound less natural. In this paper, we propose a statistical speech synthesis system based on a hidden semi-Markov model (HSMM), which can be viewed as an HMM with explicit state duration PDFs. The use of HSMMs can solve the above inconsistency because we can incorporate the state duration PDFs explicitly into both the synthesis and the training parts of the system. Subjective listening test results show that use of HSMMs improves the reported naturalness of synthesized speech.

Minimizing the Side Lobe Level (SLL) and attain highest achievable Beam Collection Efficiency (BCE) is a critical goal for Solar Power Station/Satellite (SPS). If all antennas are uniformly excited then the main beam will carry only a part of the total energy due to the higher SLL. SLL is decreased and BCE is increased by adopting edge tapering for SPS. But edge tapering is a complex technical problem for SPS. So an optimization is needed between uniform amplitude distribution and edge tapering system. We have derived a new method of edge tapering called Isosceles Trapezoidal Distribution (ITD) edge tapering. Only a small number of antennas from each side of the phased array antenna are tapered in this method. ITD edge tapering is almost uniform so it is technically better. We have compared different amplitude distribution systems; uniform, Gaussian, Dolph-Chebyshev and the newly derived ITD method. The SLL reduction in ITD is even lower than those of other kinds of edge tapering. Therefore the amount of losing power in the SLL in ITD is lower. As a result the interference level becomes lower and BCE becomes higher in this method. The higher BCE and better SLL performance than those with uniform distribution can be achieved in ITD with phase error and under unit failed condition.