A novel Uniform Discrete Multitone (DMT) transceiver is proposed, utilizing a wavelet packet based filter bank transmultiplexer in conjunction with a DMT transceiver. The proposed transceiver decomposes the channel spectrum into subbands of equal bandwidth. The objective is to minimize the bit error rate (BER), which is increased by channel-noise amplification. This noise amplification is due to the Zero-Forcing equalization (ZFE) technique. Quantization of the channel-noise amplification is presented, based on post-equalization signal-to-noise ratio (SNR) and probability of error in all subbands of the Uniform DMT system. A modified power loading algorithm is applied to allocate variable power according to subband gains. A BER performance comparison of the Uniform DMT with variable and uniform power-loading and with a conventional DMT system in a Digital Subscriber Line (DSL) channel is presented.

We propose an adaptive SLM scheme based on peak observation for PAPR reduction of OFDM signals. The proposed scheme is composed of three steps: peak scaling, sequence selection, and SLM procedures. In the first step, the peak signal samples in the IFFT outputs of the original input sequence are scaled down. In the second step, the sub-carrier positions where the power difference between the original input sequence and the FFT output of the scaled signal is large, are identified. Then, the phase sequences having the maximum number of phase-reversed sequence words only for these positions are selected. Finally, the generic SLM procedure is performed by using only the selected phase sequences for the original input sequence. Simulation results show that the proposed scheme significantly reduce the complexity in terms of IFFT and PAPR calculation than the conventional SLM, while maintaining the PAPR reduction performance.

Energy-efficiency is one of the main concerns in the wireless information dissemination system. This paper presents a wireless broadcast stream organization scheme which enables complex queries (e.g., aggregation queries) to be processed in an energy-efficient way. For efficient processing of complex queries, we propose an approach of broadcasting their pre-computed results with the data stream, wherein the way of replication of index and pre-computation results are investigated. Through analysis and experiments, we show that the new approach can achieve significant performance enhancement for complex queries with respect to the access time and tuning time.

A novel low insertion-loss and wideband microstrip bandpass filter has been designed and tested. The basic configuration of this novel dual-mode filter is a square ring resonator with direct-connected orthogonal feed lines, and dual-perturbation elements are introduced within the resonator at symmetrical location. The effects of the size of the perturbation element are studied. A new filter having wider bandwidth and transmission zeros are presented. The proposed filter responses are in good agreement with the simulations and experiments.

A scheme for a low-voltage CMOS syllabic-companding log domain filter with wide dynamic range is proposed and its prototype is presented. A nodal voltage which is fixed in a conventional filter based on the dynamically adjustable biasing (DAB) technique is adapted for change of input envelope to achieve wide dynamic range. Externally linear and time invariant (ELTI) relation between an input and an output is guaranteed by a state variable correction (SVC) circuit which is also proposed for low-voltage operation. To demonstrate the proposed scheme, a fifth-order Chebychev low-pass filter with 100-kHz cutoff frequency is designed and fabricated in a standard 0.35-µm CMOS process. The filter has a 78-dB dynamic range and consumes 200-µW power from a 0.8-V power supply.

H.264/AVC is the newest video coding standard. There are many new features in it which can be easily used for video encryption. In this paper, we propose a new scheme to do video encryption for H.264/AVC video compression standard. We define Unequal Secure Encryption (USE) as an approach that applies different encryption schemes (with different security strength) to different parts of compressed video data. This USE scheme includes two parts: video data classification and unequal secure video data encryption. Firstly, we classify the video data into two partitions: Important data partition and unimportant data partition. Important data partition has small size with high secure protection, while unimportant data partition has large size with low secure protection. Secondly, we use AES as a block cipher to encrypt the important data partition and use LEX as a stream cipher to encrypt the unimportant data partition. AES is the most widely used symmetric cryptography which can ensure high security. LEX is a new stream cipher which is based on AES and its computational cost is much lower than AES. In this way, our scheme can achieve both high security and low computational cost. Besides the USE scheme, we propose a low cost design of hybrid AES/LEX encryption module. Our experimental results show that the computational cost of the USE scheme is low (about 25% of naive encryption at Level 0 with VEA used). The hardware cost for hybrid AES/LEX module is 4678 Gates and the AES encryption throughput is about 50 Mbps.

Auto-calibration for structure and motion recovery can be used for match move where the goal is to insert synthetic 3D objects into real scenes and create views as if they were part of the real scene. However, most auto-calibration methods for multi-views utilize bundle adjustment with non-linear optimization, which requires a very good starting approximation. We propose a novel key-frame selection measurement and LMedS (Least Median of Square)-based approach to estimate scene structure and motion from image sequences captured with a hand-held camera. First, we select key-frames considering the ratio of number of correspondences and feature points, the homography error and the distribution of corresponding points in the image. Then, by using LMedS, we reject erroneous frames among the key-frames in absolute quadric estimation. Simulation results demonstrated that the proposed method can select suitable key-frames efficiently and achieve more precise camera pose estimation without non-linear optimization.

Program analysis techniques have improved steadily over the past several decades, and software obfuscation schemes have come to be used in many commercial programs. A software obfuscation scheme transforms an original program or a binary file into an obfuscated program that is more complicated and difficult to analyze, while preserving its functionality. However, the security of obfuscation schemes has not been properly evaluated. In this paper, we analyze obfuscation schemes in order to clarify the advantages of our scheme, the XOR-encoding scheme. First, we more clearly define five types of attack models that we defined previously, and define quantitative resistance to these attacks. Then, we compare the security, functionality and efficiency of three obfuscation schemes with encoding variables: (1) Sato et al.'s scheme with linear transformation, (2) our previous scheme with affine transformation, and (3) the XOR-encoding scheme. We show that the XOR-encoding scheme is superior with regard to the following two points: (1) the XOR-encoding scheme is more secure against a data-dependency attack and a brute force attack than our previous scheme, and is as secure against an information-collecting attack and an inverse transformation attack as our previous scheme, (2) the XOR-encoding scheme does not restrict the calculable ranges of programs and the loss of efficiency is less than in our previous scheme.

The performance of single-carrier (SC) transmission in a frequency-selective fading channel degrades due to a severe inter-symbol interference (ISI). Using frequency-domain equalization (FDE) based on the minimum mean square error (MMSE) criterion can improve the bit error rate (BER) performance of SC transmission. However, the residual ISI after FDE limits the performance improvement. In this paper, we propose a joint use of Tomlinson-Harashima precoding (THP) and FDE to remove the residual ISI. An approximate conditional BER analysis is presented for the given channel condition. The achievable average BER performance is evaluated by Monte-Carlo numerical computation method using the derived conditional BER. The BER analysis is confirmed by computer simulation of the signal transmission.

The analysis of multiband quantum transport simulation in double-gate metal oxide semiconductor field effects transistors (DG-MOSFETs) is performed based on a non-equilibrium Green's function (NEGF) formalism coupled self-consistently with the Poisson equation. The empirical sp3s* tight binding approximation (TBA) with nearest neighbor coupling is employed to obtain a realistic multiband structure. The effects of non-parabolic bandstructure as well as anisotropic features of Si are studied and analyzed. As a result, it is found that the multiband simulation results on potential and current profiles show significant differences, especially in higher applied bias, from those of conventional effective mass model.

A low complexity minimum mean squared error (MMSE) equalizer for orthogonal frequency division multiplexing (OFDM) systems over time-varying channels is presented. It uses a small matrix of dominant partial channel information and recursive calculation of matrix inverse to significantly reduce the complexity. Theoretical analysis and simulations results are provided to validate its significant performance or complexity advantages over the previously published MMSE equalizers.

In this paper, sampling rate selection diversity (SRSD) scheme for Direct-Sequence/Spread-Spectrum (DS/SS) is proposed. In DS/SS communication systems, oversampling may be employed to increase the signal-to-noise ratio (SNR). However, oversampling enlarges the power consumption because signal processing of the receiver has to be carried out at a higher clock rate. Higher sampling rate does not always maximize the SNR. In the proposed SRSD scheme, the power consumption can be reduced by selecting the optimum sampling rate depending on the characteristics of the channel. The proposed SRSD scheme can also reduce the BER more than the conventional oversampling scheme under certain channel conditions.

IrBurst, recently proposed by IrDA, is a high speed information transmission protocol. In this paper, a mathematical model is developed which leads to derivation of the IrBurst throughput over the IrDA protocol stack. Based on this model, we compare the performance of IrBurst and existing OBEX protocol in order to investigate the suitability of IrBurst protocol for exchange of large data blocks over high-speed IrDA links. Furthermore, the model allows the evaluation of the impact of the link layer parameters, such as window size and frame length, and physical layer parameters, such as minimum turnaround time, on system throughput for high-speed IrDA links and in the presence of transmission errors. Consequently, an effective Automatic Repeat Request (ARQ) scheme is proposed at link layer to maximize the throughput efficiency for IrBurst protocol as well as for next generation high speed IrDA links. Simulation result indicates that employment of our proposed ARQ scheme results in significant improvement of IrBurst throughput efficiency at high bit error rates.

In this paper, a new family of binary sequences of period 2n-1 with low correlation is proposed for integer n=em and even m. The new family has family size 2n+1 and maximum nontrivial correlation +1 and +1 for even and odd e respectively. Especially, for n=2m and 3m, we obtain a new family of binary sequences with maximum nontrivial correlation +1, and the obtained family is one of the binary families with best correlation among the known families with family size no less than their period 2n-1 for even n. Moreover, the correlation distribution of the new family is also determined.

Recently, we provided closed-form expressions involving two-dimensional (2-D) joint Gaussian Q-function for the symbol error rate (SER) and bit error rate (BER) of an arbitrary 2-D signal with I/Q unbalances over an additive white Gaussian noise (AWGN) channel [1]. In this letter, we extend the expressions to Nakagami-m fading channels. Using Craig representation of the 2-D joint Gaussian Q-function, we derive an exact and general expression for the error probabilities of arbitrary 2-D signaling with I/Q phase and amplitude unbalances over Nakagami-m fading channels.

In one-cell-frequency-reuse Orthogonal Frequency Division Multiple Access based Time Division Multiple Access (OF/TDMA) systems, communication is blocked by interference from adjacent cells. The most promising solution would be an adaptive modulation and coding scheme that is controlled by estimating the signal-to-interference ratio (SIR). However, there has so far been no way to accurately estimate the SIR using the spreading codes for OF/TDMA systems, because of the asynchronous fast Fourier transform (FFT). In this paper, we propose a novel SIR estimation method that uses a spread pulse-wave symbol and carrier interferometry. Moreover, to introduce multi- input multi-output systems, we modify the proposed method by allocating a different spreading code to each cell. Computer simulation confirmed that the SIR is estimated accurately even if the FFT is asynchronous. On cell boundaries, the average estimation errors that are a ratio between accurate and estimated propagation characteristics are less than 2 dB.

A semi-coherent technique for low-complexity frequency offset estimation is presented. The proposed estimation is based on discrete-time Fourier transform (DTFT). Since residual frequency offset can be compensated by channel estimation, frequency offset can be estimated only coarsely. We take advantage of the relationship between frequency resolution and accumulation period in DTFT in deriving the coarse estimator. Based on that, a novel method to balance the coherent and the non-coherent accumulation for frequency offset estimation is proposed. The proposed algorithm has low latency and complexity so that it is particularly suitable for packet traffic. The semi-coherent structure of the proposed algorithm is also scalable so that it can be used for both bursty and continuous traffic.

Normalization transform is known to be very useful for finding the overall trend of time-series data since it enables finding sequences with similar fluctuation patterns. Previous subsequence matching methods with normalization transform, however, would incur index overhead both in storage space and in update maintenance since they should build multiple indexes for supporting query sequences of arbitrary length. To solve this problem, we adopt a single-index approach in the normalization-transformed subsequence matching that supports query sequences of arbitrary length. For the single-index approach, we first provide the notion of inclusion-normalization transform by generalizing the original definition of normalization transform. To normalize a window, the inclusion-normalization transform uses the mean and the standard deviation of a subsequence that includes the window while the original transform uses those of the window itself. Next, we formally prove the correctness of the proposed normalization-transformed subsequence matching method that uses the inclusion-normalization transform. We then propose subsequence matching and index-building algorithms to implement the proposed method. Experimental results for real stock data show that our method improves performance by up to 2.52.8 times compared with the previous method.

VoIP is one of the key technologies for recent telecommunication services. In addition to the migration from the conventional PSTN to IP networks, mobile networks will follow the PSTN in moving to an IP-based infrastructure. Due to limited radio resources, the speech bitrate in mobile networks must be more strongly compressed than is true in PSTN. This will lead to a heterogeneous network environment, in which different speech codecs are employed in fixed and mobile networks. Therefore, from the viewpoint of designing and managing the QoE (Quality of Experience) of end-to-end telephony services, establishing a method to evaluate the quality of VoIP in such a heterogeneous network environment is very important. The quality of speech communication services should be discussed in subjective terms. Subjective quality assessment is time-consuming and expensive, however, so objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. To establish an objective method to evaluate the end-to-end quality of speech in a heterogeneous network environment, this paper proposes a method for estimating the end-to-end listening quality based on the quality in each individual segment. This method is very important because conventional technologies such as the E-model, which was standardized as ITU-T Recommendation G.107, cannot accurately estimate overall quality based on segmental qualities. The experimentals show that the proposed method offers better performance in terms of quality estimation than the conventional method.

Rapid advances in wireless sensor networks require routing protocols which can accommodate new types of power source and data of differing priorities. We describe a QoS-aware geographic routing scheme based on a solar-cell energy model. It exploits an algorithm (APOLLO) that periodically and locally determines the topological knowledge range (KR) of each node, based on an estimated energy budget for the following period which includes the current energy, the predicted energy consumption, and the energy expected from the solar cell. A second algorithm (PISA) runs on each node and uses its knowledge range to determine a route which meets the objectives of each priority level in terms of path delay, energy consumption and reliability. These algorithms maximize scalability and minimize memory requirements by employing a localized routing method which only uses geographic information about the host node and its adjacent neighbors. Simulation results confirm that APOLLO can determine an appropriate KR for each node and that PISA can meet the objectives of each priority level effectively.