IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences
Online ISSN : 1745-1337
Print ISSN : 0916-8508
Volume E94.A , Issue 10
Showing 1-24 articles out of 24 articles from the selected issue
Regular Section
• Kensaku FUJII, Kenji KASHIHARA, Mitsuji MUNEYASU, Masakazu MORIMOTO
Type: PAPER
Subject area: Noise and Vibration
2011 Volume E94.A Issue 10 Pages 1899-1906
Published: October 01, 2011
Released: October 01, 2011
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In this paper, we propose a method capable of shortening the distance from a noise detection microphone to a loudspeaker, which is one of important issues in the field of active noise control (ANC). In the ANC system, the secondary noise provided by the loudspeaker is required arriving at an error microphone simultaneously with the primary noise to be cancelled. However, the reverberation involved in the secondary path from the loudspeaker to the error microphone increases the secondary noise components arriving later than the primary noise. The late components are not only invalid for canceling the primary noise but also impede the cancellation. To reduce the late components, the distance between the noise detection microphone and the loud speaker is generally extended. The proposed method differently reduces the late components by forming the noise control filter, which produces the secondary noise, with the cascade connection of a non-recursive and a recursive filters. The distance can be thus shortened. On the other hand, the recursive filter is required to work stably. The proposed method guarantees the stable work by forming the recursive filter with the lattice filter whose coefficients are restricted to less than unity.
• Toshio ITO, Masanori SUGIMOTO, Hiromichi HASHIZUME
Type: PAPER
Subject area: Ultrasonics
2011 Volume E94.A Issue 10 Pages 1907-1919
Published: October 01, 2011
Released: October 01, 2011
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This paper presents and evaluates a new acoustic imaging system that uses multicarrier signals for correlation division in synthetic transmit aperture (CD-STA). CD-STA is a method that transmits uncorrelated signals from different transducers simultaneously to achieve high-speed and high-resolution acoustic imaging. In CD-STA, autocorrelations and cross-correlations in transmitted signals must be suppressed because they cause artifacts in the resulting images, which narrow the dynamic range as a consequence. To suppress the correlation noise, we had proposed to use multicarrier signals optimized by a genetic algorithm. Because the evaluation of our proposed method was very limited in the previous reports, we analyzed it more deeply in this paper. We optimized three pairs of multicarrier waveforms of various lengths, which correspond to 5th-, 6th- and 7th-order M-sequence signals, respectively. We built a CD-STA imaging system that operates in air. Using the system, we conducted imaging experiments to evaluate the image quality and resolution of the multicarrier signals. We also investigated the ability of the proposed method to resolve both positions and velocities of target scatterers. For that purpose, we carried out an experiment, in which both moving and fixed targets were visualized by our system. As a result of the experiments, we confirmed that the multicarrier signals have lower artifact levels, better axial resolution, and greater tolerance to velocity mismatch than M-sequence signals, particularly for short signals.
• Taichi YOSHIDA, Taizo SUZUKI, Seisuke KYOCHI, Masaaki IKEHARA
Type: PAPER
Subject area: Digital Signal Processing
2011 Volume E94.A Issue 10 Pages 1920-1927
Published: October 01, 2011
Released: October 01, 2011
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In this paper, we propose a two dimensional (2D) non-separable adaptive directional lifting (ADL) structure for discrete wavelet transform (DWT) and its image coding application. Although a 2D non-separable lifting structure of 9/7 DWT has been proposed by interchanging some lifting, we generalize a polyphase representation of 2D non-separable lifting structure of DWT. Furthermore, by introducing the adaptive directional filteringingto the generalized structure, the 2D non-separable ADL structure is realized and applied into image coding. Our proposed method is simpler than the 1D ADL, and can select the different transforming direction with 1D ADL. Through the simulations, the proposed method is shown to be efficient for the lossy and lossless image coding performance.
• Masahiro IWAHASHI, Hitoshi KIYA
Type: PAPER
Subject area: Digital Signal Processing
2011 Volume E94.A Issue 10 Pages 1928-1936
Published: October 01, 2011
Released: October 01, 2011
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This paper proposes a reversible two dimensional (2D) discrete wavelet transform (DWT) for lossless coding which is compatible with the irreversible 2D 9-7 DWT for lossy coding in the JPEG 2000. Since all the filters and scalings are factorized into a product of lifting steps, and signal values are rounded into integers, the proposed DWT is reversible and applicable to lossless coding of 2D signals. We replace a part of the separable 2D transfer function of the 2D DWT by a non separable 2D lifting structure, so that the number of rounding operations is decreased. We also investigate performance of the DWT under octave decomposition case and theoretically endorse it. As a result, reduction of the rounding errors due to the replacement was confirmed. It means that compatibility of the reversible DWT to the irreversible 2D 9-7 DWT is improved.
• Kuo-Hsiung TSENG, Tuo-Wen CHANG, Ming-Fu HUNG
Type: PAPER
Subject area: Systems and Control
2011 Volume E94.A Issue 10 Pages 1937-1947
Published: October 01, 2011
Released: October 01, 2011
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This study focused on three simplified models, namely (1) one set of single-phase DC-DC converter, (2) two sets of parallel connection single-phase DC-DC converter, and (3) two sets of series connection single-phase DC-DC converter. The purposes are: (1) to propose the simplification conditions and procedures for the three-phase AC-DC converter; (2) propose a set of new simplification steps for modeling, and present the examples of different three-phase AC-DC circuit topologies, detailed discussion on the simplification steps for modeling of a three-phase AC-DC converter is offered, to help people simplify and analyze the simplified model easily; (3) according to three types of simplified modeling in the three-phase AC-DC converter, this study established a useful reference for the design and analysis of the control systems of the three-phase AC-DC converter simply; (4) to acquire PWM control strategy beforehand based on PFC-Controlled property; (5) to reduce the switching loss for the PWM control strategy of the simplified model; (6) to maintain the original circuit topology and verify that the theory can extensively apply the knowledge of single-phase DC-DC converter to the simplified modeling of three-phase AC-DC converter.
• Takaaki OKUMURA, Masanori HASHIMOTO
Type: PAPER
Subject area: VLSI Design Technology and CAD
2011 Volume E94.A Issue 10 Pages 1948-1953
Published: October 01, 2011
Released: October 01, 2011
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This paper discusses how to cope with dynamic power supply noise in FF timing estimation. We first review the dependence of setup and hold times on supply voltage, and point out that setup time is more sensitive to supply voltage than hold time, and hold time at nominal voltage is reasonably pessimistic. We thus propose a procedure to estimate setup time and clock-to-Q delay taking into account given voltage drop waveforms using an equivalent DC voltage approach. Experimental results show that the proposed procedure estimates setup time and clock-to-Q delay fluctuations well with 5% and 3% errors on average.
• Hideki TAKASE, Hiroyuki TOMIYAMA, Hiroaki TAKADA
Type: PAPER
Subject area: VLSI Design Technology and CAD
2011 Volume E94.A Issue 10 Pages 1954-1964
Published: October 01, 2011
Released: October 01, 2011
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• Andry SETIAWAN, Shin-ichi NAKANO
Type: PAPER
Subject area: Algorithms and Data Structures
2011 Volume E94.A Issue 10 Pages 1965-1970
Published: October 01, 2011
Released: October 01, 2011
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In this paper we give a simple algorithm to generate all st-orientations of a given biconnected plane graph G with a designated edge (s,t) on the outer face of G. Our algorithm generates all st-orientations of G in O(n) time for each without duplications, where n is the number of vertices.
• Takeshi SUGAWARA, Naofumi HOMMA, Takafumi AOKI, Akashi SATOH
Type: PAPER
Subject area: Cryptography and Information Security
2011 Volume E94.A Issue 10 Pages 1971-1980
Published: October 01, 2011
Released: October 01, 2011
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This paper proposes an efficient scheme for concurrent error detection for hardware implementations of the block cipher AES. In the proposed scheme, the circuit component for the round function is divided into two stages, which are used alternately for encryption (or decryption) and error checking in a pipeline. The proposed scheme has a limited overhead with respect to size and speed for the following reasons. Firstly, the need for a double number of clock cycles is eliminated by virtue of the reduced critical path. Secondly, the scheme only requires minimal additional circuitry for error detection since the detection is performed by the remaining encryption (or decryption) components within the pipeline. AES hardware with the proposed scheme was designed and synthesized by using 90-nm CMOS standard cell library with various constraints. As a result, the proposed circuit achieved 1.66Gbps @ 12.9 Kgates for the compact version and 4.22Gbps @ 30.7 Kgates for the high-speed version. These performance characteristics are comparable to those of a basic AES circuit without error detection, where the overhead of the proposed scheme is estimated to be 14.5% at maximum. The proposed circuit was fabricated in the form of a chip, and its error detection performance was evaluated through experiments. The chip was tested with respect to fault injection by using clock glitch, and the proposed scheme successfully detected and reacted to all introduced errors.
• Sung-Ming YEN, Wei-Chih LIEN, Chien-Ning CHEN
Type: PAPER
Subject area: Cryptography and Information Security
2011 Volume E94.A Issue 10 Pages 1981-1990
Published: October 01, 2011
Released: October 01, 2011
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Power analysis can be used to attack many implementations of cryptosystems, e.g., RSA and ECC, and the doubling attack is a collision based power analysis performed on two chosen ciphertexts. In this paper, we introduced a modified doubling attack to threaten RSA and ECC implementations by exploiting only one chosen ciphertext of small order. To attack the RSA implementations we selected an input of order two while to attack the ECC implementations we exploited one chosen invalid point of small order on a cryptographically weak curve rather than on the original curve. We showed that several existing power analysis countermeasures for RSA and ECC implementations are still vulnerable to the proposed attack. To prevent the proposed attack, we suggested countermeasures for RSA as well as for ECC.
• Xuan GUANG, Fang-Wei FU
Type: PAPER
Subject area: Coding Theory
2011 Volume E94.A Issue 10 Pages 1991-2001
Published: October 01, 2011
Released: October 01, 2011
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In network coding, for the case that the network topology is unknown completely, random linear network coding has been proposed as an acceptable coding technique. In this paper, we define average failure probability of random linear network coding in order to characterize the performance of random network coding, and then analyze this failure probability for different known topological information of network. We obtain several upper bounds on the failure probabilities, and further show that, for some networks, these upper bounds are tight or asymptotically tight. Moreover, if the more topological information of the network is utilized, the better upper bounds are acquired.
• Marzieh DASHTI, Mir GHORAISHI, Katsuyuki HANEDA, Jun-ichi TAKADA, Keni ...
Type: PAPER
Subject area: Spread Spectrum Technologies and Applications
2011 Volume E94.A Issue 10 Pages 2002-2012
Published: October 01, 2011
Released: October 01, 2011
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This paper proposes a method for setting the threshold for ultra-wide-band (UWB) threshold-based ranging in indoor scenarios. The optimum threshold is derived based on the full analysis of the ranging error, which is equivalent to the probability of correct detection of first arriving signal in time-based ranging techniques. It is shown that the probability of correct detection is a function of first arriving signal, which has variations with two independent distributions. On the one hand, the first arriving signal varies in different positions with the same range due to multipath interference and on the other, it is a function of distance due to free space path-loss. These two distributions are considered in the derivation of the ranging error, based on which the optimum threshold is obtained. A practical method to derive this threshold is introduced based on the standard channel model. Extensive Monte Carlo simulations, ray-tracing simulations and ranging measurements confirm the analysis and the superior performance of the proposed threshold scheme.
• Jia SU, Yiqing HUANG, Lei SUN, Shinichi SAKAIDA, Takeshi IKENAGA
Type: PAPER
Subject area: Image
2011 Volume E94.A Issue 10 Pages 2013-2021
Published: October 01, 2011
Released: October 01, 2011
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With the increasing demand of high video quality and large image size, adaptive interpolation filter (AIF) addresses these issues and conquers the time varying effects resulting in increased coding efficiency, comparing with recent H.264 standard. However, currently most AIF algorithms are based on either frame level or macroblock (MB) level, which are not flexible enough for different video contents in a real codec system, and most of them are facing a severe time consuming problem. This paper proposes a content based coarse to fine AIF algorithm, which can adapt to video contents by adding different filters and conditions from coarse to fine. The overall algorithm has been mainly made up by 3 schemes: frequency analysis based frame level skip interpolation, motion vector modeling based region level interpolation, and edge detection based macroblock level interpolation. According to the experiments, AIF are discovered to be more effective in the high frequency frames, therefore, the condition to skip low frequency frames for generating AIF coefficients has been set. Moreover, by utilizing the motion vector information of previous frames the region level based interpolation has been designed, and Laplacian of Gaussian based macroblock level interpolation has been proposed to drive the interpolation process from coarse to fine. Six 720p and six 1080p video sequences which cover most typical video types have been tested for evaluating the proposed algorithm. The experimental results show that the proposed algorithm reduce total encoding time about 41% for 720p and 25% for 1080p sequences averagely, comparing with Key Technology Areas (KTA) Enhanced AIF algorithm, while obtains a BDPSNR gain up to 0.004 and 3.122 BDBR reduction.
• Seokjin LEE, Sang Ha PARK, Koeng-Mo SUNG
Type: LETTER
Subject area: Engineering Acoustics
2011 Volume E94.A Issue 10 Pages 2022-2026
Published: October 01, 2011
Released: October 01, 2011
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In this paper, an on-line nonnegative matrix factorization (NMF) algorithm for acoustic signal processing systems is developed based on the recursive least squares (RLS) method. In order to develop the on-line NMF algorithm, we reformulate the NMF problem into multiple least squares (LS) normal equations, and solve the reformulated problems using RLS methods. In addition, we eliminate the irrelevant calculations based on the NMF model. The proposed algorithm has been evaluated with a well-known dataset used for NMF performance evaluation and with real acoustic signals; the results show that the proposed algorithm performs better than the conventional algorithm in on-line applications.
• Xiao ZHOU, Fang YANG, Jian SONG
Type: LETTER
Subject area: Noise and Vibration
2011 Volume E94.A Issue 10 Pages 2027-2030
Published: October 01, 2011
Released: October 01, 2011
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To reduce the error of channel estimation caused by noise, a novel noise suppression method based on the degree of confidence is proposed in this paper. The false alarm and false dismissal probabilities, corresponding to noise being taken as part of channel impulse response (CIR) and part of the CIR being mis-detected as noise, respectively, are also investigated. A false alarm reduction method is therefore presented to reduce the false alarms in the estimated CIR while the mis-detection ratio still remains low. Simulation results show the effectiveness of the proposed method.
• Jae-Hun CHOI, Joon-Hyuk CHANG, Dong Kook KIM, Suhyun KIM
Type: LETTER
Subject area: Speech and Hearing
2011 Volume E94.A Issue 10 Pages 2031-2034
Published: October 01, 2011
Released: October 01, 2011
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In this paper, we propose a spectral difference approach for noise power estimation in speech enhancement. The noise power estimate is given by recursively averaging past spectral power values using a smoothing parameter based on the current observation. The smoothing parameter in time and frequency is adjusted by the spectral difference between consecutive frames that can efficiently characterize noise variation. Specifically, we propose an effective technique based on a sigmoid-type function in order to adaptively determine the smoothing parameter based on the spectral difference. Compared to a conventional method, the proposed noise estimate is computationally efficient and able to effectively follow noise changes under various noise conditions.
• Hui CHEN, Qun WAN, Hongyang CHEN, Tomoaki OHTSUKI
Type: LETTER
Subject area: Digital Signal Processing
2011 Volume E94.A Issue 10 Pages 2035-2038
Published: October 01, 2011
Released: October 01, 2011
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A new direction of arrival (DOA) estimation method is introduced with arbitrary array geometry when uncorrelated and coherent signals coexist. The DOAs of uncorrelated signals are first estimated via subspace-based high resolution DOA estimation technique. Then a matrix that only contains the information of coherent signals can be formulated by eliminating the contribution of uncorrelated signals. Finally a subspace block sparse reconstruction approach is taken for DOA estimations of the coherent signals.
• Taehwan KIM, Keunsung BAE
Type: LETTER
Subject area: Digital Signal Processing
2011 Volume E94.A Issue 10 Pages 2039-2042
Published: October 01, 2011
Released: October 01, 2011
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This paper deals with underwater target classification using synthesized active sonar signals. Firstly, we synthesized active sonar returns from a 3D highlight model of underwater targets using the ray tracing algorithm. Then, we applied a multiaspect target classification scheme based on a hidden Markov model to classify them. For feature extraction from the synthesized sonar signals, a matching pursuit algorithm was used. The experimental results depending on the number of observations and signal-to-noise ratios are presented with our discussions.
• Masahiro HAYASHI
Type: LETTER
Subject area: Reliability, Maintainability and Safety Analysis
2011 Volume E94.A Issue 10 Pages 2043-2047
Published: October 01, 2011
Released: October 01, 2011
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Although unavailability and failure frequency are important reliability measures for designing telecommunications networks, having two different reliability measures causes difficulties when their evaluation results contradict each other. To overcome this difficulty, for determining a single reliability measure is proposed. Market share, which is an important management index, was used and the effect of each reliability measure on this index was studied. If we the effect of one of the reliability measures is small enough to be ignored, then the other one is chosen as the single reliability measure. The procedure of the proposed statistical analysis method and test results suggesting that failure frequency is promising as a single reliability measure are discussed.
• Leida LI, Jianying ZHANG, Ajith ABRAHAM
Type: LETTER
Subject area: Cryptography and Information Security
2011 Volume E94.A Issue 10 Pages 2048-2052
Published: October 01, 2011
Released: October 01, 2011
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This letter presents a new image watermarking scheme using Polar Sine Transform (PST), a new kind of orthogonal moment defined on a circular domain. The PSTs are easy to compute and have no numerical stability problem, thus are more suitable for watermarking. In the proposed method, the PSTs are modified according to the binary watermark bits, producing a compensation image. The watermarked image is obtained by adding the compensation image to the original image directly. Simulation results show the advantages of the proposed scheme in terms of both watermark capacity and watermark robustness.
Type: LETTER
Subject area: Cryptography and Information Security
2011 Volume E94.A Issue 10 Pages 2053-2058
Published: October 01, 2011
Released: October 01, 2011
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In this paper, we study how exploiting multiple differential characteristics with a common initial difference and different output differences improves the complexity of differential cryptanalysis attack. We call such an approach Multiple Differential Cryptanalysis. We describe such an attack rigorously by studying the probability distribution of multiple differential characteristics and giving an attack algorithm based on LLR statistic. We also present a statistical analysis on the attack complexity based on LLR probabilistic technique. Our analysis shows that the data complexity of the proposed attack decreases as the number of characteristics increases. We do an experiment with the described method to show its improvements through cryptanalyzing a reduced round PRESENT block cipher with 5 rounds.
• Xiaoyan ZHANG, Qichun WANG, Bin WANG, Haibin KAN
Type: LETTER
Subject area: Cryptography and Information Security
2011 Volume E94.A Issue 10 Pages 2059-2062
Published: October 01, 2011
Released: October 01, 2011
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In algebraic attack on stream ciphers based on LFSRs, the secret key is found by solving an overdefined system of multivariate equations. There are many known algorithms from different point of view to solve the problem, such as linearization, relinearization, XL and Gröbner Basis. The simplest method, linearization, treats each monomial of different degrees as a new variable, and consists of $\\sum_{i=1}^{d}{n \\choose i}$ variables (the degree of the system of equations is denoted by d). Thus it needs at least $\\sum_{i=1}^{d}{n \\choose i}$ equations, i.e. keystream bits to recover the secret key by Gaussian reduction or other. In this paper we firstly propose a concept, called equivalence of LFSRs. On the basis of it, we present a constructive method that can solve an overdefined system of multivariate equations with less keystream bits by extending the primitive polynomial.