This paper discusses the behavior of the second-order modes (Hankel singular values) of linear continuous-time systems under variable transformations with positive-real functions. That is, given a transfer function H(s) and its second-order modes, we analyze the second-order modes of transformed systems H(F(s)), where 1/F(s) is an arbitrary positive-real function. We first discuss the case of lossless positive-real transformations, and show that the second-order modes are invariant under any lossless positive-real transformation. We next consider the case of general positive-real transformations, and reveal that the values of the second-order modes are decreased under any general positive-real transformation. We achieve the derivation of these results by describing the controllability/observability Gramians of transformed systems, with the help of the lossless positive-real lemma, the positive-real lemma, and state-space formulation of transformed systems.

This paper proposes several cepstral statistics compensation and normalization algorithms which alleviate the effect of additive noise on cepstral features for speech recognition. The algorithms are simple yet efficient noise reduction techniques that use online-constructed pseudo-stereo codebooks to evaluate the statistics in both clean and noisy environments. The process yields transformations for both clean speech cepstra and noise-corrupted speech cepstra, or for noise-corrupted speech cepstra only, so that the statistics of the transformed speech cepstra are similar for both environments. Experimental results show that these codebook-based algorithms can provide significant performance gains compared to results obtained by using conventional utterance-based normalization approaches. The proposed codebook-based cesptral mean and variance normalization (C-CMVN), linear least squares (LLS) and quadratic least squares (QLS) outperform utterance-based CMVN (U-CMVN) by 26.03%, 22.72% and 27.48%, respectively, in relative word error rate reduction for experiments conducted on Test Set A of the Aurora-2 digit database.

Recently, there has been extensive research on resource allocation schemes for multicast services that would satisfy the requirements of multimedia traffic. Although several schemes have been proposed to improve the performance of individual multicast groups, it is not easy to achieve both throughput efficiency and user fairness. In this study, we propose a new multicast scheduling scheme for achieving proportional fair (PF) allocation in wireless cellular systems. The basic idea of PF is to schedule the user whose corresponding instantaneous channel quality is the highest relative to the average channel condition over a given time scale. We first extend the PF metric to the extent that the scheduler can reflect the user's varying channel gain, and fairness, not only in the unicast case, but also in multicast transmissions. A multicast PF scheme maximizes the summation of the logarithmic average rate of all multicasting users. Thus, it improves the fairness to mobile users when compared to max-rate allocation, because the logarithmic rate gives more weight to lower rate users, while achieving high throughput. Moreover, the proposed scheme is less complex than max-rate allocation.

We suggest a dual thresholds method for the dynamic bandwidth allocation in EPON. In the suggested method, a buffer in ONU has two thresholds and ONU generates a normal request and a greedy request based on the two thresholds. Also, OLT estimates the overall traffic load and grants the greedy request when estimated traffic is light. We study upstream channel resource wastage and show the suggested method decreases the upstream channel resource wastage. Using simulation, we validate the dual thresholds method is superior to the existing methods in the mean delay.

This paper proposes a performance model for design of pipelined analog-to-digital converters (ADCs). This model includes the effect of overdrive voltage on the transistor, slewing of the operational amplifier, multi-bit structure of multiplying digital to analog converter (MDAC) and technology scaling. The conversion frequency of ADC is improved by choosing the optimum overdrive voltage of the transistor, an important consideration at smaller design rules. Moreover, multi-bit MDACs are faster than the single-bit MDACs when slewing occurs during the step response. The performance model of pipelined ADC shown in this paper is attractive for the optimization of the ADC's performances.

The historical review of Taiwan's researching activities on the features of PECVD grown SiOx are also included to realize the performance of Si nanocrystal based MOSLED made by such a Si-rich SiOx film with embedded Si nanocrystals on conventional Si substrate. A surface nano-roughened Si substrate with interfacial Si nano-pyramids at SiOx/Si interface are also reviewed, which provide the capabilities of enhancing the surface roughness induced total-internal-reflection relaxation and the Fowler-Nordheim tunneling based carrier injection. These structures enable the light emission and extraction from a metal-SiOx-Si MOSLED.

For the purpose of reducing the quantization noise and power consumption of UWB-OFDM transceiver, a new time domain-based interpolator and decimator structure is proposed to realize five-bit D/A and A/D converters in the five-bit 128-tone sigma-delta modulation (SDM) UWB-OFDM transceiver. The five-bit 128-tone SDM UWB-OFDM transceiver using time domain-based interpolator and decimator in place of time spreader and de-spreader can obtain time-domain spread spectrum processing gain and reduce quantization noise simultaneously. The structure of the five-bit 128-tone SDM A/D converter, which employs 32 parallel analog SDM circuits without up-sampling, is designed. Simulation results demonstrate that BER of the proposed five-bit 128-tone SDM D/A and A/D converters based on time domain-based interpolator and decimator scheme can satisfy the performance requirements of the five-bit 128-tone SDM UWB-OFDM transceiver for the QPSK, 16-QAM and 64-QAM modulations.

We propose a simple asynchronous UWB (Ultra Wide Band) position location algorithm with low complexity, power consumption and processing delay. In the proposed algorithm, only a single RTTX (Round-Trip Transmission) of UWB pulses is utilized based on the ToA (Time of Arrival) principle. Hence, the proposed algorithm decreases power consumption and processing delay as compared to the basic ToA based on triple RTTXs. Moreover, unlike the TDoA (Time Difference of Arrival) algorithm, the proposed algorithm can perform the position location with low complexity since it does not require strict synchronization between multiple beacons. Simulation results using IEEE 802.15.4a UWB channel models reveal that the proposed algorithm achieves closely comparable position location performance of the basic ToA and TDoA algorithms.

In this paper, we discuss the collision search for hash functions, mainly in terms of their advanced message modification. The advanced message modification is a collision search tool based on Wang et al.'s attacks. Two advanced message modifications have previously been proposed: cancel modification for MD4 and MD5, and propagation modification for SHA-0. In this paper, we propose a new concept of advanced message modification, submarine modification. As a concrete example combining the ideas underlying these modifications, we apply submarine modification to the collision search for SHA-0. As a result, we show that this can reduce the collision search attack complexity from 239 to 236 SHA-0 compression operations.

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.

We first model the formal security model of multisignature scheme following that of group signature scheme. Second, we prove that the following three probabilistic multisignature schemes based on a trapdoor permutation have tight security; PFDH (probabilistic full domain hash) based multisignature scheme (PFDH-MSS), PSS (probabilistic signature scheme) based multisignature scheme (PSS-MSS), and short signature PSS based multisignature scheme (S-PSS-MSS). Third, we give an optimal proof (general result) for multisignature schemes, which derives the lower bound for the length of random salt. We also estimate the upper bound for the length in each scheme and derive the optimal length of a random salt. Two of the schemes are promising in terms of security tightness and optimal signature length. In appendix, we describe a multisignature scheme using the claw-free permutation and discuss its security.

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.

In this paper, an analytical model is proposed to compute the optimal number of clusters that minimizes the energy consumption of multi-hop wireless sensor networks. In the proposed analytical model, the average hop count between a general node (GN) and its nearest clusterhead (CH) is obtained assuming a uniform distribution. How the position of the sink impacts the optimal number of clusters is also discussed. A numerical simulation is carried out to validate the proposed model in various network environments.

In this paper we analyze the characteristics of vehicle mobility and propose a novel Mobility Prediction Progressive Routing (MP2R) protocol for Inter-Vehicle Communication (IVC) that is based on cross-layer design. MP2R utilizes the additional gain provided by the directional antennas to improve link quality and connectivity; interference is reduced by the directional transmission. Each node learns its own position and speed and that of other nodes, and performs position prediction. (i) With the predicted progress and link quality, the forwarding decision of a packet is locally made, just before the packet is actually transmitted. In addition the load at the forwarder is considered in order to avoid congestion. (ii) The predicted geographic direction is used to control the beam of the directional antenna. The proposed MP2R protocol is especially suitable for forwarding burst traffic in highly mobile environments. Simulation results show that MP2R effectively reduces Packet Error Ratio (PER) compared with both topology-based routing (AODV [1], FSR [2]) and normal progressive routing (NADV [18]) in the IVC scenarios.

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.

In this paper, a fiber with two inhomogeneous sector holes around the core is proposed, and propagation characteristics of polarization maintaining region and single-polarization region are numerically analyzed by circular Fourier expansion method. In each case of the single-polarization region and the polarization maintaining region, a fiber is designed so as to satisfy the zero total dispersion at wavelength of 1.55 µm. Then, the single-polarization bandwidth for the single-polarization region and the modal birefringence for the polarization maintaining region are examined as the specific characteristics in each region. In addition, the power concentrating into the core region and distributions of Poynting vector is also discussed.

A concurrent model of computation and a language based on the model for bit-level operation are useful for developing asynchronous and concurrent programs compositionally, which frequently use bit-level operations. Some examples are programs for video games, hardware emulation (including virtual machines), and signal processing. However, few models and languages are optimized and oriented to bit-level concurrent computation. We previously developed a visual programming language called A-BITS for bit-level concurrent programming. The language is based on a dataflow-like model that computes using processes that provide serial bit-level operations and FIFO buffers connected to them. It can express bit-level computation naturally and develop compositionally. We then devised a concurrent computation model called APEC (Asynchronous Program Elements Connection) for bit-level concurrent computation. This model enables precise and formal expression of the process of computation, and a notion of primitive program elements for controlling and operating can be expressed synthetically. Specifically, the model is based on a notion of uniform primitive processes, called primitives, that have three terminals and four ordered rules at most, as well as on bidirectional communication using vehicles called carriers. A new notion is that a carrier moving between two terminals can briefly express some kinds of computation such as synchronization and bidirectional communication. The model's properties make it most applicable to bit-level computation compositionally, since the uniform computation elements are enough to develop components that have practical functionality. Through future application of the model, our research may enable further research on a base model of fine-grain parallel computer architecture, since the model is suitable for expressing massive concurrency by a network of primitives.

In this paper, an architecture of MIMO mesh network which avoids co-channel interference and supplies link multiplexing simultaneously, namely MIMO spatial spectrum sharing, is proposed. As a MIMO transmission scheme, linear (such as zero-forcing) and nonlinear (such as dirty paper coding and successive interference cancellation) MIMO algorithm are developed for the proposed mesh network. It is found from numerical analysis that the proposed MIMO mesh network achieves significantly higher channel capacity than that of conventional mesh networks.

This paper describes a smart thermal sensing chip with an integrated vertical bipolar transistor sensor, a Sigma Delta Modulator (SDM), a Micro-Control Unit (MCU), and a bandgap reference voltage generator for biomedical application by using 0.18 µm CMOS process. The npn bipolar transistors with the Deep N-Well (DNW) instead of the pnp bipolar transistor is first adopted as the sensor for good isolation from substrate coupling noise. In addition to data compression, Micro-Control Unit (MCU) plays an important role for executing auto-calibration by digitally trimming the bipolar sensor in parallel to save power consumption and to reduce feedback complexity. It is different from the present analog feedback calibration technologies. Using one sensor, instead of two sensors, to create two differential signals in 180phase difference input to SDM is also a novel design of this work. As a result, in the range of 0 to 80 or body temperature (375), the inaccuracy is less than 0.1 or 0.05 respectively with one-point calibration after packaging. The average power consumption is 268.4 µW with 1.8 V supply voltage.

A technique for suppressing the clipping noise of an analogue-to-digital converter (ADC) is proposed to realize a cognitive radio transceiver that offers high sensitivity carrier-sensing. When a large bandwidth cognitive radio transceiver performs carrier-sensing, it must receive a radio wave that includes many primary user transmissions. The radio wave may have high peak-to-average power ratio (PAPR) and clipping noise may be generated. Clipping noise becomes an obstacle to the achievement of high-sensitivity carrier-sensing. In the proposed technique, the original values of the samples clipped by an ADC are estimated by interpolation. Polynomial spline interpolation to the clipped signal is performed in the first step, and then SINC function interpolation is applied to the spline interpolated signal. The performance was evaluated using the signals with various PAPR. It has been found that suppression performance has a dependency on the number of samples clipped at once rather than on PAPR. Although there is an upper limit for the number of samples clipped at once that can be compensated with high accuracy, about 20 dB suppression of clipping noise was achieved with the medium degree of clipping.