NAND-based block devices such as memory cards and solid-state drives embed a flash translation layer (FTL) to emulate the standard block device interface and its features. The overall performance of these devices is determined mainly by the efficiency of the FTL scheme, so intensive research has been performed to improve the average performance of the FTL scheme. However, its worst-case performance has rarely been considered. The present study aims to improve the worst-case performance without affecting the average performance. The central concept is to distribute the garbage collection cost, which is the main source of performance fluctuations, over multiple requests. The proposed scheme comprises three modules: i) anticipated partial log block merging to distribute the garbage collection time; ii) reclaiming clean pages by moving valid pages to bound the worst-case garbage collection time, instead of performing repeated block merges; and iii) victim selection based on the valid page count in a victim log and the required clean page count to avoid subsequent garbage collections. A trace-driven simulation showed that the worst-case performance was improved up to 1,300% using the proposed garbage collection scheme. The average performance was also similar to that of the original scheme. This improvement was achieved without additional memory overheads.

Amplification characteristics of the signal and the noise in the semiconductor optical amplifier (SOA), without facet mirrors for the intensity modulated light, are theoretically analyzed and experimentally confirmed. We have found that the amplification factor of the temporarily varying intensity component is smaller than that of the continuous wave (CW) component, but increases up to that of the CW component in the high frequency region in the SOA. These properties are very peculiar in the SOA, which is not shown in conventional electronic devices and semiconductor lasers. Therefore, the relative intensity noise (RIN), which is defined as ratio of the square value of the intensity fluctuation to that of the CW power can be improved by the amplification by the SOA. On the other hand, the signal to the noise ratio (S/N ratio) defined for ratio of the square value of the modulated signal power to that of the intensity fluctuation have both cases of the degradation and the improvement by the amplification depending on combination of the modulation and the noise frequencies. Experimental confirmations of these peculiar characteristics are also demonstrated.

High efficiency video coding (HEVC) is a video compression standard that outperforms the predecessor H.264/AVC by doubling the compression efficiency. To enhance the intra prediction accuracy, 35 intra prediction modes were used in the prediction units (PUs), with partition sizes ranging from 4 × 4 to 64 × 64 in HEVC. However, the manifold prediction modes dramatically increase the encoding complexity. This paper proposes a fast mode- and depth-decision algorithm based on edge detection and reconfiguration to alleviate the large computational complexity in intra prediction with trivial degradation in accuracy. For mode decision, we propose pixel gradient statistics (PGS) and mode refinement (MR). PGS uses pixel gradient information to assist in selecting the prediction mode after rough mode decision (RMD). MR uses the neighboring mode information to select the best PU mode (BPM). For depth decision, we propose a partition reconfiguration algorithm to replace the original partitioning order with a more reasonable structure, by using the smoothness of the coding unit as a criterion in deciding the prediction depth. Smoothness detection is based on the PGS result. Experiment results show that the proposed method saves about 41.50% of the original processing time with little degradation (BD bitrate increased by 0.66% and BDPSNR decreased by 0.060dB) in the coding gain.

In this paper, we investigate unitary precoder design for multiple-input multiple-output (MIMO) multicasting, where multiple common data streams are sent to a group of users. Assuming that zero-forcing decision feedback equalizers (ZF-DFE) are adopted at the receiver side, we can convert the multicast channel into multiple parallel subchannels. To improve the receiving quality of all data streams, we focus on maximizing the minimal signal-to-noise ratio (SNR) of all data streams. To effectively handle this non-convex optimization problem, we first consider the special case of two data streams and derive the closed-form solution of the SNR vectors for both subchannels. Based on these results, a gradient-based iterative algorithm is developed for the proposed precoder design. For the general case, a Givens rotation-based iterative algorithm is proposed, where at each iteration the original problem of unitary precoder design is transformed into a dual-stream subproblem. Hence it can be solved efficiently by the gradient-based iterative algorithm. Finally, simulation results are presented to demonstrate the outstanding performance of the proposed design.

We have investigated the microwave-detected photoconductivity responses from the amorphous In--Ga--Zn--O (a-IGZO) thin films. The time constant extracted by the slope of the slow part of the reflectivity signals are correlated with TFT performances. We have evaluated the influences of the sputtering conditions on the quality of a-IGZO thin film, as well as the influences of gate insulation films and annealing conditions, by comparing the TFT characteristics with the microwave photoconductivity decay ($mu$-PCD). It is concluded that the $mu$-PCD is a promising method for in-line process monitoring for the IGZO-TFTs fabrication.

A fully-integrated double-channel 5-Gb/s/ch 2:1 serializer array is designed and fabricated in a standard 0.18-$mu $m CMOS technology, which can be easily expanded to any even-number-channel array, e.g. 12 channels, by means of arrangement in a parallel manner. Besides two conventional half-rate 2:1 serializers, both phase-locked loop and delay-locked loop techniques are employed locally to deal with the involved clocking-related issues, which make the serializer array self-contained, compact and automatic. The system architecture, circuit and layout designs are discussed and analyzed in detail. The chip occupies a die area of 673,$mu $m$, imes ,$667,$mu $m with a core width of only 450,$mu $m. Measurement results show that it works properly without a need for additional clock channels, reference clocks, off-chip tuning, external components, and so on. From a single supply of 1.8,V, a power of 200,mW is consumed and a single-ended swing of above 300,mV for each channel is achieved.

This study presents a proposal for space-saving design of built-in antennas for handset terminals based on the concept of requisite design antenna volume. By investigating the relation between antenna input characteristic and electric near-field around the antenna element and surrounding components inside the terminal, and then evaluating the requisite design antenna volume, we propose the most effective deployment for both the antenna and surrounding components. The results show that our simple proposal can help reduced, by about 17% and 31.75%, the space that the antenna element actually requires at least for stable operation inside the terminal, in the single-band designs for the cellular 2GHz band (1920-2170MHz) and 800MHz band (830-880MHz), respectively. In the dual-band design, we verify that it can reduce, the antenna space by about 35.18%, and completely cover the two above cellular bands with good antenna performance.

Wireless communications for the control of industrial equipments need to send a large amount of short packets frequently and to improve frame efficiency. The OFDM frame of wireless local area networks has short preambles that are used for timing synchronization and coarse frequency offset estimation. As the short preambles are repeated in a time domain, they occupies subcarriers intermittently. Therefore, in this paper, a new frame format with OFDM modulation in which data symbols are orthogonally multiplexed with the preamble symbols in the frequency domain is proposed. Two preamble sequences that are based on an IEEE802.11g short preamble sequence and a Zadoff-Chu sequence are examined. The ratio of transmission powers between the pilot subcarrier and the data subcarrier is also varied. The timing synchronization probability with those sequences has been evaluated on different channel models. It is shown through the experiment that the synchronization performance is almost the same as that without data multiplexing at Es/N0 of more than 8dB.

Hands-free communications between cellular phones must be robust enough to withstand echo-path variation, and highly nonlinear echoes must be suppressed at low cost, when acoustic echo cancellation or suppression is applied to them. This paper proposes a spectrum-selective nonlinear echo suppression (SS-ES) approach as a solution to these issues. SS-ES is characterized by the selection of either a spectrum of the residual signal from an adaptive filter or a spectrum of the sending input signal depending on the amount of linear echo cancellation in an adaptive filter. Compared to conventional methods, the objective evaluation results of the SS-ES approach show an improvement of approximately 0.8-2.2dB, 0.23-2.39dB, and 0.26-0.50 in average echo return loss enhancement (ERLE), average root-mean-square log-spectral distortion (RMS-LSD), and the perceptual evaluation of speech quality (PESQ) value, respectively, under echo-path variation and double-talk conditions.

Due to the high speed mobile environment, the aeronautical Cognitive Radio (CR) communications base on the channel with the time-variant stochastic non-continuous spectrum. The traditional fading channel models, such as Rayleigh, Rice, Nakagami-m and multipath channel models, can not describe the whole property of the channels of CR communications. In this letter, the statistical channel modeling scheme for aeronautical CR communications is proposed with a M/M/s(1) queuing model, which properly describes the random spectrum occupation of the primary users, i.e. aircrafts in aeronautical communications. The proposed channel model can be easily utilized in the channel simulation to testify the validity and efficiency of the aeronautical CR communications.

In this paper, we propose a parameter estimation method using Volterra kernels for the nonlinear IIR filters, which are used for the linearization of closed-box loudspeaker systems. The nonlinear IIR filter, which originates from a mirror filter, employs nonlinear parameters of the loudspeaker system. Hence, it is very important to realize an appropriate estimation method for the nonlinear parameters to increase the compensation ability of nonlinear distortions. However, it is difficult to obtain exact nonlinear parameters using the conventional parameter estimation method for nonlinear IIR filter, which uses the displacement characteristic of the diaphragm. The conventional method has two problems. First, it requires the displacement characteristic of the diaphragm but it is difficult to measure such tiny displacements. Moreover, a laser displacement gauge is required as an extra measurement instrument. Second, it has a limitation in the excitation signal used to measure the displacement of the diaphragm. On the other hand, in the proposed estimation method for nonlinear IIR filter, the parameters are updated using simulated annealing (SA) according to the cost function that represents the amount of compensation and these procedures are repeated until a given iteration count. The amount of compensation is calculated through computer simulation in which Volterra kernels of a target loudspeaker system is utilized as the loudspeaker model and then the loudspeaker model is compensated by the nonlinear IIR filter with the present parameters. Hence, the proposed method requires only an ordinary microphone and can utilize any excitation signal to estimate the nonlinear parameters. Some experimental results demonstrate that the proposed method can estimate the parameters more accurately than the conventional estimation method.

In this paper, we investigate the minimum-weight codewords of array LDPC codes C(m,q), where q is an odd prime and m ≤ q. Using some analytical approaches, the lower bound on the number of minimum-weight codewords of C(m,q) given by Kaji (IEEE Int. Symp. Inf. Theory, June/July 2009) is proven to be tight for m = 4 and q > 19. In other words, C(4,q) has 4q2(q-1) minimum-weight codewords for all q > 19. In addition, we show some interesting universal properties of the supports of generators of minimum-weight codewords of the code C(4,q)(q > 19).

This paper proposes a new algorithm to encode the spectral envelope for G.729.1 more accurately. It applies the normalized least-mean- square (NLMS) algorithm to each subband energy of the modified discrete cosine transform (MDCT) in the time-domain alias cancellation (TDAC) of G.729.1. By utilizing the estimation error of subband energies by means of NLMS, allocated bit reduction for spectral envelope coding is achieved. The saved bits are then reused to improve the spectral envelope estimation and thus enhance the sound quality. Experimental results confirm that the proposed algorithm improves the sound quality under both clean and packet loss conditions.

In this letter, we formally present the definition of KDM-CCA1 security in public key setting, which falls in between the existing KDM-CPA and KDM-CCA2 security. We also prove that if a public key encryption scheme is CCA1 secure and has the properties of secret-key multiplication (or addition) homomorphism, and conditioned plaintext-restorability, then it is KDM-CCA1 secure w.r.t. two ensembles of functions that had been used in [15],[17], respectively. For concrete scheme, we show that the (tailored) Damgård's Elgamal scheme achieves this KDM-CCA1 security based on different assumptions.

We focus on a characteristic which is specific to the quantum repeater protocol. In the quantum repeater protocol, quantum states which are generated by the protocol do not depend on receivers. Therefore, we can reuse EPR pairs which are generated before a change of a receiver for the quantum repeater protocol after the change. The purpose of reusing is advancing the finishing time of sharing EPR pairs, which is not equal to increasing the fidelity. In this paper, we construct concrete methods of reusing EPR pairs and analyze the effectiveness of reusing EPR pairs. Besides, we derive conditions in which reusing EPR pairs is effective.

We propose TagFlow, a data plane mechanism for classification in Software-Defined Networking (SDN). We first argue that simple field-matching proposals of current SDN APIs are not efficient and flexible enough and then propose a tag based classification mechanism as an alternative. Moreover, we propose user-defined actions as an improvement over current hardcoded actions in SDN APIs. Our experiments show TagFlow forwarding is almost 40% faster than OpenFlow. Furthermore, our user-defined actions at SDN southbound are thousands of times faster that equivalent northbound implementations in the literature.

Bandwidth aggregation (BAG) techniques have been researched for many years in an efforts to enhance throughput for multi-homed streaming service. However, despite of the considerable attention being devoted towards energy-efficient communications, the power efficiency for BAG has not been considered yet. To improve the power efficiency in multi-homed streaming service, this paper proposes Power Minimized Rate Allocation Scheme (PMRAS) with optimal rate allocation at each interface while guaranteeing an allowable packet loss rate. In developing PMRAS, we first formulate a power consumption model based on the network interface state (i.e. active and idle state). We adopt a Lagrangian algorithm to solve the convex optimization problem of power consumption. The performance results gained from a numerical analysis and simulations (NS-2) reveal that the proposed scheme offers superior performance over the existing rate allocation scheme for BAG with guaranteed required quality of service.

The traffic load of cellular networks varies in both time and spatial domains, causing many base stations (BS) to be under-utilized. Assisted by cell zooming, dynamic BS sleep control is considered as an effective way to improve energy efficiency during low traffic hours. Therefore, how densely the BSs should be deployed with cell zooming and BS sleeping is an important issue. In this paper, we explore the energy-optimal cellular network planning problem with dynamic BS sleeping and cell zooming for the cases in which traffic is uniformly distributed in space but time-varying. To guarantee the quality of multi-class services, an approximation method based on Erlang formula is proposed. Extensive simulations under our predefined scenarios show that about half of energy consumption can be saved through dynamic BS sleeping and power control. Surprisingly, the energy-optimal BS density we obtained is larger than the one without considering BS sleeping. In other words, deploying more BSs may help to save energy if dynamic BS sleeping is executed.

In this paper, we propose demodulators for the Golden and Alamouti codes in amplify-and-forward (AF) cooperative communication with one relay. The proposed demodulators output exact log likelihood ratios (LLRs) with recursion based on the Jacobian logarithm. The cooperative system with the proposed demodulator for the Golden code has the benefit of efficient data transmission, while the system for the Alamouti code has low demodulation complexity. Quantitative analyses of computational complexity of the proposed demodulators are conducted. The transmission performance for various relay location and power settings is evaluated on cooperative orthogonal frequency division multiplexing (OFDM)-based wireless local area network (LAN) systems. In evaluations, the optimal relay location and power settings are found. The cooperative system with the proposed demodulators for the Golden and Alamouti codes offers 1.5 and 1.9 times larger areas where 10.8 and 5.4Mbit/s can be obtained than a non-cooperative (direct) system in a typical office environment, respectively.

In this paper, we propose a novel Carrier Frequency Offset (CFO) estimation and compensation algorithm applicable to Software Defined Radio (SDR). A two stage estimation algorithm has been proposed as a concatenation of two algorithms namely Modified Maximum Likelihood Data Aided (MMLDA) coarse frequency estimation and sample by sample residual CFO estimation. The second stage tracks the residual offset on sample by sample basis for the whole burst without using preamble. Simulation results are given for Stanford University Interim (SUI) channels to demonstrate the effectiveness of the proposed algorithm in multipath fading channel. The proposed algorithm shows better performance than the conventional two stage algorithms, even for large frequency offsets. The proposed algorithm has been implemented in software on TMS320C64x+ Digital Signal Processor (DSP) core and verified by comparing with simulation results.