IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences - recent issues

Special Section on Acoustic Scene Analysis and Reproduction

Makino, S. — 2008-06-16

Calculating Inverse Filters for Speech Dereverberation

MIYOSHI, M., DELCROIX, M., KINOSHITA, K. — 2008-06-16

Speech dereverberation is one of the most difficult tasks in acoustic signal processing. Of the various problems involved in this task, this paper highlights "over-whitening," which flattens the characteristics of recovered speech. This distortion sometimes happens when inverse filters are directly calculated from microphone signals. This paper reviews two studies related to this problem. The first study shows the possibility of compensating for such over-whitening to achieve precise speech-dereverberation. The second study presents a new approach for approximating the original speech by removing the effect of late reflections from observed reverberant speech.

Localization Model of Synthesized Sound Image Using Precedence Effect in Sound Field Reproduction Based on Wave Field Synthesis

KIMURA, T., YAMAKATA, Y., KATSUMOTO, M., KAKEHI, K. — 2008-06-16

Although it is very important to conduct listening tests when constructing a practical sound field reproduction system based on wave field synthesis, listening tests are very expensive. A localization model of synthesized sound images that predicts the results of listening tests is proposed. This model reduces the costs of constructing a reproduction system because it makes it possible to omit the listening tests. The proposed model uses the precedence effect and predicts the direction of synthesized sound images based on the inter-aural time difference. A comparison of the results predicted by the proposed model and the localized results of listening tests shows that the model accurately predicts the localized results.

Auditory Artifacts due to Switching Head-Related Transfer Functions of a Dynamic Virtual Auditory Display

OTANI, M., HIRAHARA, T. — 2008-06-16

Auditory artifacts due to switching head-related transfer functions (HRTFs) are investigated, using a software-implemented dynamic virtual auditory display (DVAD) developed by the authors. The DVAD responds to a listener's head rotation using a head-tracking device and switching HRTFs to present a highly realistic 3D virtual auditory space to the listener. The DVAD operates on Windows XP and does not require high-performance computers. A total system latency (TSL), which is the delay between head motion and the corresponding change of the ear input signal, is a significant factor of DVADs. The measured TSL of our DVAD is about 50 ms, which is sufficient for practical applications and localization experiments. Another matter of concern is the auditory artifact in DVADs caused by switching HRTFs. Switching HRTFs gives rise to wave discontinuity of synthesized binaural signals, which can be perceived as click noises that degrade the quality of presented sound image. A subjective test and excitation patterns (EPNs) analysis using an auditory filter are performed with various source signals and HRTF spatial resolutions. The results of the subjective test reveal that click noise perception depends on the source signal and the HRTF spatial resolution. Furthermore, EPN analysis reveals that switching HRTFs significantly distorts the EPNs at the off signal frequencies. Such distortions, however, are masked perceptually by broad-bandwidth source signals, whereas they are not masked by narrow-bandwidth source signals, thereby making the click noise more detectable. A higher HRTF spatial resolution leads to smaller distortions. But, depending on the source signal, perceivable click noises still remain even with 0.5-degree spatial resolution, which is less than minimum audible angle (1 degree in front).

Rapid Compensation of Temperature Fluctuation Effect for Multichannel Sound Field Reproduction System

YAI, Y., MIYABE, S., SARUWATARI, H., SHIKANO, K., TATEKURA, Y. — 2008-06-16

In this paper, we propose a computationally efficient method of compensating temperature for the transaural stereo. The conventional method can be used to estimate the change in impulse responses caused by the fluctuation of temperature with high accuracy. However, the large amount of computation required makes real-time implementation difficult. Focusing on the fact that the amount of compensation depends on the length of the impulse response, we reduce the computation required by segmenting the impulse response. We segment the impulse responses in the time domain and estimate the effect of temperature fluctuation for each of the segments. By joining the processed segments, we obtain the compensated impulse response of the whole length. Experimental results show that the proposed method can reduce the computation required by a factor of nine without degradation of the accuracy.

A Two-Microphone Noise Reduction Method in Highly Non-stationary Multiple-Noise-Source Environments

LI, J., AKAGI, M., SUZUKI, Y. — 2008-06-16

In this paper, we propose a two-microphone noise reduction method to deal with non-stationary interfering noises in multiple-noise-source environments in which the traditional two-microphone algorithms cannot function well. In the proposed algorithm, multiple interfering noise sources are regarded as one virtually integrated noise source in each subband, and the spectrum of the integrated noise is then estimated using its virtual direction of arrival. To do this, we suggest a direction finder for the integrated noise using only two microphones that performs well even in speech active periods. The noise spectrum estimate is further improved by integrating a single-channel noise estimation approach and then subtracted from that of the noisy signal, finally enhancing the desired target signal. The performance of the proposed algorithm is evaluated and compared with the traditional algorithms in various conditions. Experimental results demonstrate that the proposed algorithm outperforms the traditional algorithms in various conditions in terms of objective and subjective speech quality measures.

Robust Frequency Domain Acoustic Echo Cancellation Filter Employing Normalized Residual Echo Enhancement

SHIMAUCHI, S., HANEDA, Y., KATAOKA, A. — 2008-06-16

We propose a new robust frequency domain acoustic echo cancellation filter that employs a normalized residual echo enhancement. By interpreting the conventional robust step-size control approaches as a statistical-model-based residual echo enhancement problem, the optimal step-size introduced in the most of conventional approaches is regarded as optimal only on the assumption that both the residual echo and the outlier in the error output signal are described by Gaussian distributions. However, the Gaussian-Gaussian mixture assumption does not always hold well, especially when both the residual echo and the outlier are speech signals (known as a double-talk situation). The proposed filtering scheme is based on the Gaussian-Laplacian mixture assumption for the signals normalized by the reference input signal amplitude. By comparing the performances of the proposed and conventional approaches through the simulations, we show that the Gaussian-Laplacian mixture assumption for the normalized signals can provide a better control scheme for the acoustic echo cancellation.

Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming

OSAKO, K., MORI, Y., TAKAHASHI, Y., SARUWATARI, H., SHIKANO, K. — 2008-06-16

We propose a new algorithm for the blind source separation (BSS) approach in which independent component analysis (ICA) and frequency subband beamforming interpolation are combined. The slow convergence of the optimization of the separation filters is a problem in ICA. Our approach to resolving this problem is based on the relationship between ICA and null beamforming (NBF). The proposed method consists of the following three parts: (I) a frequency subband selector part for learning ICA, (II) a frequency domain ICA part with direction-of-arrivals (DOA) estimation of sound sources, and (III) an interpolation part in which null beamforming constructed with the estimated DOA is used. The results of the signal separation experiments under a reverberant condition reveal that the convergence speed is superior to that of the conventional ICA-based BSS methods.

Sound Reproduction System Robust against Environmental Variation by Switching Control Band Range

TATEKURA, Y., WATANABE, T. — 2008-06-16

A robust multichannel sound reproduction system that utilizes the relationship between the width of the actual control area and the control frequency of the control points is proposed. The reproduction accuracy of a conventional sound reproduction system is reduced by room environment variations when fixed inverse filter coefficients are used. This tendency becomes more significant when control points are arranged more closely. To resolve this problem, the frequency control band at every control point is switched to avoid degrading the reproduced sound in low frequencies, so the pass band range of the control points at both ears is only high-range. That of the other control points is the entire control range. Numerical simulation with real environmental data showed that improvement of the reproduction accuracy is about 6.1 dB on average, even with a temperature fluctuation of 5°C as an environmental variation in the listening room.

Special Section on Image Media Quality

Hangai, S. — 2008-06-16

The Interaction of Art, Technology and Customers in Picture Making

MCCANN, J. J., MIYAKE, Y. — 2008-06-16

Human interest in pictures dates back to 14,000 BC. Pictures can be drawn by hand or imaged by optical means. Over time pictures have changed from being rare and unique to ubiquitous and common. They have changed from treasures to transients. This paper summarizes many picture technologies, and discusses their dynamic range, their color and tone scale rendering. This paper discusses the interactions between advances in technology and the interests of its users over time. It is the combination of both technology and society's usage that has shaped imaging since its beginning and continues to do so.

Image and Video Quality Assessment Using LCD: Comparisons with CRT Conditions

TOURANCHEAU, S., LE CALLET, P., BARBA, D. — 2008-06-16

In this paper, the impact of display on quality assessment is addressed. Subjective quality assessment experiments have been performed on both LCD and CRT displays. Two sets of still images and two sets of moving pictures have been assessed using either an ACR or a SAMVIQ protocol. Altogether, eight experiments have been led. Results are presented and discussed, some differences are pointed out. Concerning moving pictures, these differences seem to be mainly due to LCD moving artefacts such as motion blur. LCD motion blur has been measured objectively and with psycho-physics experiments. A motion-blur metric based on the temporal characteristics of LCD can be defined. A prediction model have been then designed which predict the differences of perceived quality between CRT and LCD. This motion-blur-based model enables the estimation of perceived quality on LCD with respect to the perceived quality on CRT. Technical solutions to LCD motion blur can thus be evaluated on natural contents by this mean.

Image Quality Assessment for Color Halftone Images Based on Color Structural Similarity

LEE, J., HORIUCHI, T. — 2008-06-16

This paper proposes a new color halftone image quality assessment method based upon the color structural similarity measure with considering the human visual characteristics. To include the color visual characteristics, we carry out the color filtering for each luminance, red-green, and blue-yellow channels. Then, we apply the color structural similarity measure to the color filtered images, which are the reference image and the halftoned image, to evaluate the localized structural difference. By considering those characteristics, in this paper, the assessment of the color halftone images can be realized. We apply the proposed measure to the various kinds of color halftone images and confirm that the proposed measure can give reasonable results compared with the results by subjective evaluation.

A Development of the TFT-LCD Image Defect Inspection Method Based on Human Visual System

OH, J.-H., YUN, B.-J., KIM, S.-Y., PARK, K.-H. — 2008-06-16

The TFT-LCD image has non-uniform brightness that is the major difficulty of finding the visible defect called Mura in the field. To facilitate Mura detection, background signal shading should level off and Mura signal must be amplified. In this paper, Mura signal amplification and background signal flattening method is proposed based on human visual system (HVS). The proposed DC normalized contrast sensitivity function (CSF) is used for the Mura signal amplification and polynomial regression (PR) is used to level off the background signal. In the enhanced image, tri-modal thresholding segmentation technique is used for finding Dark and White Mura at the same time. To select reliable defect, falsely detected invisible region is eliminated based on Weber's Law. By the experimental results of artificially generated 1-d signal and TFT-LCD image, proposed algorithm has novel enhancement results and can be applied to real automated inspection system.

A Masking Model for Motion Sharpening Phenomenon in Video Sequences

FUJIBAYASHI, A., BOON, C. S. — 2008-06-16

In this paper, we show that motion sharpening phenomenon can be explained as a form of visual masking for a special case where a video sequence is composed of alternate frames with different level of sharpness. A frame of higher sharpness behaves to mask the ambiguity of a subsequent frame of lower sharpness and hence preserves the perceptive quality of the whole sequence. Borrowing the mechanism for visual masking, we formulated a quantitative model for deriving the minimum spatial frequency conditions which preserves the subjective quality of the frames being masked. The quantitative model takes into account three fundamental properties of the video signals, namely the size of motion, average luminance and the power of each frequency components. The psychophysical responses towards the changes of these properties are obtained through subjective assessment tests using video sequences of simple geometrical patterns. Subjective experiments on natural video sequences show that more than 75% of viewers could make no distinction between the original sequence and the one processed using the quantitative model.

Identification of Piecewise Linear Uniform Motion Blur

PATANUKHOM, K., NISHIHARA, A. — 2008-06-16

A motion blur identification scheme is proposed for non-linear uniform motion blurs approximated by piecewise linear models which consist of more than one linear motion component. The proposed scheme includes three modules that are a motion direction estimator, a motion length estimator and a motion combination selector. In order to identify the motion directions, the proposed scheme is based on a trial restoration by using directional forward ramp motion blurs along different directions and an analysis of directional information via frequency domain by using a Radon transform. Autocorrelation functions of image derivatives along several directions are employed for estimation of the motion lengths. A proper motion combination is identified by analyzing local autocorrelation functions of non-flat component of trial restored results. Experimental examples of simulated and real world blurred images are given to demonstrate a promising performance of the proposed scheme.

Wavelet Based Multipurpose Color Image Watermarking by Using Dual Watermarks with Human Vision System Models

TSAI, M.-J., LIN, C.-W. — 2008-06-16

In this study, we propose a complete architecture based on digital watermarking techniques to solve the issue of copyright protection and authentication for digital contents. We apply visible and semi-fragile watermarks as dual watermarks where visible watermarking is used to establish the copyright protection and semi-fragile watermarking authenticates and verifies the integrity of the watermarked image. In order to get the best tradeoff between the embedding energy of watermark and the perceptual translucence for visible watermark, the composite coefficients using global and local characteristics of the host and watermark images in the discrete wavelet transform (DWT) domain is considered with Human Vision System (HVS) models. To achieve the optimum noise reduction of the visibility thresholds for HVS in DWT domain, the contrast-sensitive function (CSF) and noise visible function (NVF) of perceptual model is applied which characterizes the global and local image properties and identifies texture and edge regions to determine the optimal watermark locations and strength at the watermark embedding stage. In addition, the perceptual weights according to the basis function amplitudes of DWT coefficients is fine tuned for the best quality of perceptual translucence in the design of the proposed watermarking algorithm. Furthermore, the semi-fragile watermark can detect and localize malicious attack effectively yet tolerate mild modifications such as JPEG compression and channel additive white Gaussian noise (AWGN). From the experimental results, our proposed technique not only improves the PSNR values and visual quality than other algorithms but also preserves the visibility of the watermark visible under various signal processing and advanced image recovery attacks.

Factors of Incomplete Adaptation for Color Reproduction Considering Subjective White Point Shift for Varying Illuminant

LEE, S.-H., LEE, M.-H., SOHNG, K.-I. — 2008-06-16

In this paper, we investigated the effect of chromaticity and luminance of surround to decide subject neutral white, and conducted a mathematical model of adapting degree for environment. Factors for adapting degree consist of two parts, adapting degree of ambient chromaticity and color saturation. These can be applied to color appearance models (CAM), actually improve the performance of color matching of CAM, hence would produce the method of image reproduction to general display systems.

A Study about the Relationship between Frame Quality and Single Video Quality

KAWAYOKE, Y., HORITA, Y. — 2008-06-16

Digital video encapsulates the time series of a frame (still) images, where overall video quality can be obtained by using the quality of each frame image and the temporal information between the frame image. Coding of video produces degradation of these two types of information. These degradations can be classified as spatial degradation (static degradation) of a frame images and temporal degradation between frame image (dynamic degradation). In the framework of video quality evaluation it is necessary to consider those degradations, because their contents are strongly interdependable and quantification is problematic for these degradations. Therefore, the development of an objective video quality assessment method for single video quality requires to investigate how much static degradation and dynamic degradation affect single video quality. In this research, single video quality was predicted highly accuratly by using frame quality as static degradation and frame rate information as dynamic degradation.

A New Matrix Method for Reconstruction of Band-Limited Periodic Signals from the Sets of Integrated Values

PETROVIC, P. — 2008-06-16

This paper presents a new method for reconstruction of trigonometric polynomials, a specific class of bandlimited signals, from a number of integrated values of input signals. It is applied in signal reconstruction, spectral estimation, system identification, as well as in other important signal processing problems. The proposed method of processing can be used for precise rms measurements of periodic signal (or power and energy) based on the presented signal reconstruction. Based on the value of the integral of the original input (analogue) signal, with a known frequency spectrum but unknown amplitudes and phases, a reconstruction of its basic parameters is done by the means of derived analytical and summarized expressions. Subsequent calculation of all relevant indicators related to the monitoring and processing of ac voltage and current signals is provided in this manner. Computer simulation demonstrating the precision of these algorithms. We investigate the errors related to the signal reconstruction, and provide an error bound around the reconstructed time domain waveform.

Artificial Spiking Neurons and Analog-to-Digital-to-Analog Conversion

TORIKAI, H., TANAKA, A., SAITO, T. — 2008-06-16

This paper studies encoding/decoding function of artificial spiking neurons. First, we investigate basic characteristics of spike-trains of the neurons and fix parameter value that can minimize variation of spike-train length for initial value. Second we consider analog-to-digital encoding based upon spike-interval modulation that is suitable for simple and stable signal detection. Third we present a digital-to-analog decoder in which digital input is applied to switch the base signal of the spiking neuron. The system dynamics can be simplified into simple switched dynamical systems and precise analysis is possible. A simple circuit model is also presented.

Self-Organizing Map with False-Neighbor Degree between Neurons for Effective Self-Organization

MATSUSHITA, H., NISHIO, Y. — 2008-06-16

In the real world, it is not always true that neighboring houses are physically adjacent or close to each other. in other words, "neighbors" are not always "true neighbors." In this study, we propose a new Self-Organizing Map (SOM) algorithm, SOM with False-Neighbor degree between neurons (called FN-SOM). The behavior of FN-SOM is investigated with learning for various input data. We confirm that FN-SOM can obtain a more effective map reflecting the distribution state of input data than the conventional SOM and Growing Grid.

Low-Complexity Parallel Systolic Montgomery Multipliers over GF(2m) Using Toeplitz Matrix-Vector Representation

LEE, C.-Y. — 2008-06-16

In this paper, a generalized Montgomery multiplication algorithm in GF(2^m) using the Toeplitz matrix-vector representation is presented. The hardware architectures derived from this algorithm provide low-complexity bit-parallel systolic multipliers with trinomials and pentanomials. The results reveal that our proposed multipliers reduce the space complexity of approximately 15% compared with an existing systolic Montgomery multiplier for trinomials. Moreover, the proposed architectures have the features of regularity, modularity, and local interconnection. Accordingly, they are well suited to VLSI implementation.

Fast Custom Instruction Identification Algorithm Based on Basic Convex Pattern Model for Supporting ASIP Automated Design

ZHAO, K., BIAN, J., DONG, S., SONG, Y., GOTO, S. — 2008-06-16

To improve the computation efficiency of the application specific instruction-set processor (ASIP), a strategy of hardware/software collaborative design is usually utilized. In this process, the auto-customization of specific instruction set has always been a key part to support the automated design of ASIP. The key issue of this problem is how to effectively reduce the huge exponential exploration space in the instruction identification process. To address this issue, we first formulate it as a feasible sub-graph enumeration problem under multiple constraints, and then propose a fast instruction identification algorithm based on a new model called basic convex pattern (BCP). The kernel technique in this algorithm is the transformation from the graph exploration to the formula-based computations. The experimental results have indicated that the proposed algorithm has a distinct reduction in the execution time.

A Construction of Lossy Source Code Using LDPC Matrices

MIYAKE, S., MURAMATSU, J. — 2008-06-16

Research into applying LDPC code theory, which is used for channel coding, to source coding has received a lot of attention in several research fields such as distributed source coding. In this paper, a source coding problem with a fidelity criterion is considered. Matsunaga et al. and Martinian et al. constructed a lossy code under the conditions of a binary alphabet, a uniform distribution, and a Hamming measure of fidelity criterion. We extend their results and construct a lossy code under the extended conditions of a binary alphabet, a distribution that is not necessarily uniform, and a fidelity measure that is bounded and additive and show that the code can achieve the optimal rate, rate-distortion function. By applying a formula for the random walk on lattice to the analysis of LDPC matrices on Z_q, where q is a prime number, we show that results similar to those for the binary alphabet condition hold for Z_q, the multiple alphabet condition.

On the Minimum Weight of Simple Full-Length Array LDPC Codes

SUGIYAMA, K., KAJI, Y. — 2008-06-16

We investigate the minimum weights of simple full-length array LDPC codes (SFA-LDPC codes). The SFA-LDPC codes are a subclass of LDPC codes, and constructed algebraically according to two integer parameters p and j. Mittelholzer and Yang et al. have studied the minimum weights of SFA-LDPC codes, but the exact minimum weights of the codes are not known except for some small p and j. In this paper, we show that the minimum weights of the SFA-LDPC codes with j = 4 and j = 5 are upper-bounded by 10 and 12, respectively, independent from the prime number p. By combining the results with Yang's lower-bound limits, we can conclude that the minimum weights of the SFA-LDPC codes with j = 4 and p >7 are exactly 10 and those of the SFA-LDPC codes with j = 5 are 10 or 12.

A Combination of Adaptive Equalizer and LMS-RAKE Combining Scheme for DS-UWB System

TOH, K. B., TACHIKAWA, S. — 2008-06-16

This paper proposes a combination of adaptive equalizer and Least Mean Square-RAKE (LMS-RAKE) combining scheme receiver system for Direct Sequence-Ultra Wideband (DS-UWB) multipath channel model. The main purpose of the proposed system is to overcome the performance degradation for UWB transmission due to the occurrence of Inter-Symbol Interference (ISI) during high speed transmission of ultra short pulses in a multipath channel. The proposed system improves the system performance by mitigating the multipath effect using LMS-RAKE receiver and suppressing the ISI effect with the adaptive equalizer. Simulation results verify that significant equalization gain can be obtained by the proposed system especially in UWB multipath channel models such as channel CM3 and channel CM4 that suffered severe ISI effect.

A Pre-Emptive Horizontal Channel Borrowing and Vertical Traffic Overflowing Channel Allocation Scheme for Overlay Networks

ZHAO, F.-m., JIANG, L.-g., HE, C. — 2008-06-16

In this paper, a channel allocation scheme is studied for overlay wireless networks to optimize connection-level QoS. The contributions of our work are threefold. First, a channel allocation strategy using both horizontal channel borrowing and vertical traffic overflowing (HCB-VTO) is presented and analyzed. When all the channels in a given macro-cell are used, high-mobility real-time handoff requests can borrow channels from adjacent homogeneous cells. In case that the borrowing requests fail, handoff requests may also be overflowed to heterogeneous cells, if possible. Second, high-mobility real-time service is prioritized by allowing it to pre-empt channels currently used by other services. And third, to meet the high QoS requirements of some services and increase the utilization of radio resources, certain services can be transformed between real-time services and non-real-time services as necessary. Simulation results demonstrate that the proposed schemes can improve system performance.

A New Method for Estimating Intra Prediction Mode in H.264/AVC

KIM, D.-Y., KIM, D.-K., LEE, Y.-L. — 2008-06-16

In order to reduce spatial redundancies, the H.264/AVC Intra coding provides nine directional prediction modes including DC prediction for every 4 x 4 block, but it needs a lot of overhead bits to represent the nine directional prediction modes for every 4 x 4 block. To compress the directional mode bits efficiently, the most probable mode is estimated by using the correlation between the prediction mode of spatially adjacent blocks and that of the current block. In this paper, a new method for estimating the most probable mode is proposed by using the directional information of the prediction mode of the adjacent blocks. Experimental results show that the proposed method is able to achieve a coding gain of about 0.2 dB on average at low bit rate.

Optimizing Markov Model Parameters for Asynchronous Impulsive Noise over Broadband Power Line Communication Network

TAN, T.-H., HUANG, S.-Y., CHANG, C.-S., HUANG, Y.-F. — 2008-06-16

A statistical model based on a partitioned Markov-chains model has previously been developed to represent time domain behavior of the asynchronous impulsive noise over a broadband power line communication (PLC) network. However, the estimation of its model parameters using the Simplex method can easily trap the final solution at a local optimum. This study proposes an estimation scheme based on the genetic algorithm (GA) to overcome this difficulty. Experimental results show that the proposed scheme yields estimates that more closely match the experimental data statistics.

Conditional Converge Cast

INOUE, D., TANAKA, K. — 2008-06-16

In this paper, we introduce a new notion of conditional converge cast (CCC), by adding the conditional property to converge cast. A CCC protocol with predicate Q is a three-party protocol which involves two senders S₀ and S₁ and a receiver R. S₀ owns a secret x and a message m₀, so does S₁ with y and m₁. In a protocol, S₀ and S₁ send their messages to R in a masked form. R obtains the message depending on the value of Q(x,y), i.e. R obtains m₀ if Q(x,y)=0, or m₁ otherwise. The secrets, x and y, are not revealed to R or the other sender, and Q(x,y) is not revealed to S₀ and S₁. In addition to the formulation, we propose a concrete scheme for conditional converge cast with the "equality" predicate.

Bit-Depth Scalable Video Coding Based on H.264/AVC

PARK, S., RAO, K.R. — 2008-06-16

A bit-depth scalability is proposed in an adaptive way based on modified inter-layer predictions of the spatial scalability. A simple prediction for high dynamic range (HDR) sequences is implemented to reduce the redundancy of the residual signals between the base layer which contains low dynamic range (LDR) sequences and the enhancement layer which contains HDR sequences by using scaling and offset values.

Identity-Based Encryptions with Tight Security Reductions to the BDH Problem

NISHIOKA, M. — 2008-05-16

We present IND-ID-CPA secure identity-based encryption (IBE) schemes with tight reductions to the bilinear Diffie-Hellman (BDH) problem. Since the methods for obtaining IND-ID-CCA secure schemes from IND-ID-CPA secure schemes with tight reductions are already known, we can consequently obtain IND-ID-CCA secure schemes with tight reductions to the BDH problem. Our constructions are based on IBE schemes with tight reductions to the list bilinear Diffie-Hellman (LBDH) problem, and the schemes are converted to those with tight reductions to the BDH problem. Interestingly, it can be shown that there exists a black box construction, in which the former IBE schemes are given as black boxes. Our constructions are very simple and reasonably efficient.

A Digital Signature Scheme Based on NP-Complete Lattice Problems

HAYASHI, S., TADA, M. — 2008-05-16

In [13], we proposed new decision problems related to lattices, and proved their NP-completeness. In this paper, we present a new public-key identification scheme and a digital signature scheme based on one of the problems in [13]. We also prove the security of our schemes under certain assumptions, and analyze the efficiency of ours.

A Simple Adaptive Algorithm for Principle Component and Independent Component Analysis

SHIN, H.-C., KIM, H.-N., SONG, W.-J. — 2008-05-16

In this letter we propose a simple adaptive algorithm which solves the unit-norm constrained optimization problem. Instead of conventional parameter norm based normalization, the proposed algorithm incorporates single parameter normalization which is computationally much simpler. The simulation results illustrate that the proposed algorithm performs as good as conventional ones while being computationally simpler.

A Closed Form Solution to L2-Sensitivity Minimization of Second-Order State-Space Digital Filters

YAMAKI, S., ABE, M., KAWAMATA, M. — 2008-05-16

This paper proposes a closed form solution to L₂-sensitivity minimization of second-order state-space digital filters. Restricting ourselves to the second-order case of state-space digital filters, we can express the L₂-sensitivity by a simple linear combination of exponential functions and formulate the L₂-sensitivity minimization problem by a simple polynomial equation. As a result, the L₂-sensitivity minimization problem can be converted into a problem to find the solution to a fourth-degree polynomial equation of constant coefficients, which can be algebraically solved in closed form without iterative calculations.

A Global Stability Analysis of a Class of Nolinear Time-Delay Systems Using Continued Fraction Property

CHOI, J.-Y. — 2008-05-16

We consider a class of nonlinear time delay systems with time-varying delays, and achieve a time delay independent sufficient condition for the global asymptotic stability. The sufficient condition is proved by constructing a continued fraction that represents the lower and upper bound variations of the system trajectory along the current of time, and showing that the continued fraction converges to the equilibrium point of the system. The simulation results show the validity of the sufficient condition, and illustrate that the sufficient condition is a close approximation to the unknown necessary and sufficient condition for the global asymptotic stability.

Efficient Calculation of the Transition Matrix in a Max-Plus Linear State-Space Representation

GOTO, H. — 2008-05-16

This research considers an efficient method for calculating the transition matrix in an MPL (Max-Plus Linear) state-space representation. This matrix can be generated by applying the Kleene star operator to an adjacency matrix. The proposed method, based on the idea of a topological sort in graph theory and block splitting, is able to calculate the transition matrix efficiently.

Extended MPEG Video Format for Efficient Dynamic Voltage Scaling

BANG, K., BANG, S.-Y., CHUNG, E.-Y. — 2008-05-16

We present an extended MPEG video format for efficient Dynamic Voltage Scaling (DVS). DVS technique has been widely researched, but the execution time variation of a periodic task (i.e. MPEG decoding) is still a challenge to be tackled. Unlike previous works, we focus on the data (video stream) rather than the execution code to overcome such limitation. The proposed video format provides the decoding costs of frames to help the precise prediction of their execution times at client machines. The experimental results show that the extended format only increases the data size less than 1% by adding about 10 bits representing the decoding cost of each frame. Also, a DVS technique adjusted for the proposed format achieves 90% of efficiency compared to the oracle case, while keeping the run time overhead of the technique negligible.

Binary Constant Weight Codes Based on Cyclic Difference Sets

LI, N., ZENG, X., HU, L. — 2008-05-16

Based on cyclic difference sets, sequences with two-valued autocorrelation can be constructed. Using these constructed sequences, two classes of binary constant weight codes are presented. Some codes proposed in this paper are proven to be optimal.

Performance of Multiband OFDM Systems with Extra Diversity

KIM, J.-H., KIM, S.-T., SONG, H.-K., YOU, Y.-H. — 2008-05-16

This letter presents the performance of ultra-wideband multi-band orthogonal frequency division multiplexing (UWB MB-OFDM) systems with an extra diversity. To fully obtain diversity gain in the current MB-OFDM system when a time-domain spreading (TDS) is adopted, two consecutive OFDM symbols are designed to be cyclic shifted against each other. Simulation results indicate that the MB-OFDM system using additional frequency diversity outperforms conventional MB-OFDM system.

Improved Fine Frequency Synchronization Method for OFDM-Based DRM

YOU, Y.-H., KIM, S.-T., LEE, K.-T., SONG, H.-K. — 2008-05-16

In this letter, a robust pilot-assisted synchronization scheme is proposed for estimation of residual frequency offset (RFO) in OFDM-based digital radio mondiale (DRM) system. The RFO estimator uses the gain reference pilots mainly reserved for the channel tracking in the DRM standard. To demonstrate the efficiency of the proposed RFO estimator, comparisons are made with the conventional RFO estimator using the frequency reference pilots in terms of mean square error (MSE) performance.

Special Section on Selected Papers from the 20th Workshop on Circuits and Systems in Karuizawa

Abe, M. — 2008-04-14

Design of Class DE Amplifier with Nonlinear Shunt Capacitances for Any Output Q

EZAWA, T., SEKIYA, H., YAHAGI, T. — 2008-04-14

This paper investigates the design curves of the class DE amplifier with the nonlinear shunt capacitances for any output Q and any grading coefficient m of the diode junction in the MOSFET. The design curves are derived by the numerical calculation using Spice. The results of this paper have two important meanings. Firstly, it is clarified that the nonlinearities of the shunt capacitances affect the design curves of the class DE amplifier, especially, for low output Q. Moreover, the supply voltage is a quite important parameter to design the class DE amplifier with the nonlinear shunt capacitances. Secondly, it is also clarified that the numerical design tool using Spice, which is proposed by authors, can be applied to the derivation of the design curves. This shows the possibility of the algorithm to be a powerful tool for the analysis of the class E switching circuits. The waveforms from Spice simulations denote the validity of the design curves.

Hardware Neural Network for a Visual Inspection System

CHUN, S., HAYAKAWA, Y., NAKAJIMA, K. — 2008-04-14

The visual inspection of defects in products is heavily dependent on human experience and instinct. In this situation, it is difficult to reduce the production costs and to shorten the inspection time and hence the total process time. Consequently people involved in this area desire an automatic inspection system. In this paper, we propose a hardware neural network, which is expected to provide high-speed operation for automatic inspection of products. Since neural networks can learn, this is a suitable method for self-adjustment of criteria for classification. To achieve high-speed operation, we use parallel and pipelining techniques. Furthermore, we use a piecewise linear function instead of a conventional activation function in order to save hardware resources. Consequently, our proposed hardware neural network achieved 6GCPS and 2GCUPS, which in our test sample proved to be sufficiently fast.

Recalling Temporal Sequences of Patterns Using Neurons with Hysteretic Property

SVEHOLM, J., HAYAKAWA, Y., NAKAJIMA, K. — 2008-04-14

Further development of a network based on the Inverse Function Delayed (ID) model which can recall temporal sequences of patterns, is proposed. Additional advantage is taken of the negative resistance region of the ID model and its hysteretic properties by widening the negative resistance region and letting the output of the ID neuron be almost instant. Calling this neuron limit ID neuron, a model with limit ID neurons connected pairwise with conventional neurons enlarges the storage capacity and increases it even further by using a weightmatrix that is calculated to guarantee the storage after transforming the sequence of patterns into a linear separation problem. The network's tolerance, or the model's ability to recall a sequence, starting in a pattern with initial distortion is also investigated and by choosing a suitable value for the output delay of the conventional neuron, the distortion is gradually reduced and finally vanishes.

Application of Correlation-Based Regression Analysis for Improvement of Power Distribution Network

HAGIWARA, S., UEZONO, T., SATO, T., MASU, K. — 2008-04-14

Stochastic approaches for effective power distribution network optimization are proposed. Considering node voltages obtained using dynamic voltage drop analysis as sample variables, multi-variate regression is conducted to optimize clock timing metrics, such as clock skew or jitter. Aggregate correlation coefficient (ACC) which quantifies connectivity between different chip regions is defined in order to find a possible insufficiency in wire connections of a power distribution network. Based on the ACC, we also propose a procedure using linear regression to find the most effective region for improving clock timing metrics. By using the proposed procedure, effective fixing point were obtained two orders faster than by using brute force circuit simulation.

An Evaluation Method of the Number of Monte Carlo STA Trials for Statistical Path Delay Analysis

IMAI, M., SATO, T., NAKAYAMA, N., MASU, K. — 2008-04-14

We present an evaluation method for estimating the lower bound number of Monte Carlo STA trials required to obtain at least one sample which falls within top-k % of its parent population. The sample can be used to ensure that target designs are timing-error free with a predefined probability using the minimum computational cost. The lower bound number is represented as a closed-form formula which is general enough to be applied to other verifications. For validation, Monte Carlo STA was carried out on various benchmark data including ISCAS circuits. The minimum number of Monte Carlo runs determined using the proposed method successfully extracted one or more top-k % delay instances.

Novel Architecture of Feedforward Second-Order Multibit {Delta}{Sigma}AD Modulator

2008-04-14

This paper proposes novel feedforward architecture of the second-order multibit AD modulator with single DAC-feedback topology. The AD modulator realizes high resolution by oversampling and noise shaping techniques. However, its SNDR (Signal to Noise and Distortion Ratio) is limited by the dynamic range of the input signal and non-idealities of circuit building blocks, particularly by the harmonic distortion in amplifier circuits. A full feedforward AD modulator structure has the signal transfer function of unity under ideal circumstances, which means that the signal swings through the loop filter become smaller compared with a feedbacked AD modulator. Therefore, the harmonic distortion generated inside the loop filter can be significantly reduced in the feedforward structure because the effect of non-idealities in amplifiers can be suppressed when signal swing is small. Moreover, the reduction of the internal signal swings also relaxes output swing requirements for amplifiers with low supply voltage. However, in conventional feedforward AD modulator, an analog adder is needed before quantizer, and especially in a multibit modulator, an additional amplifier is necessary to realize the summation of feedforward signals, which leads to extra chip area and power dissipation. In this paper, we propose a novel architecture of a feedforward AD modulator which realizes the summation of feedforward signals without additional amplifier. The proposed architecture is functionally equivalent to the conventional one but with smaller chip area and lower power dissipation. We conducted MATLAB and SPICE simulations to validate the proposed architecture and modulator circuits.

A High-Speed Design of Montgomery Multiplier

FAN, Y., IKENAGA, T., GOTO, S. — 2008-04-14

With the increase of key length used in public cryptographic algorithms such as RSA and ECC, the speed of Montgomery multiplication becomes a bottleneck. This paper proposes a high speed design of Montgomery multiplier. Firstly, a modified scalable high-radix Montgomery algorithm is proposed to reduce critical path. Secondly, a high-radix clock-saving dataflow is proposed to support high-radix operation and one clock cycle delay in dataflow. Finally, a hardware-reused architecture is proposed to reduce the hardware cost and a parallel radix-16 design of data path is proposed to accelerate the speed. By using HHNEC 0.25 µm standard cell library, the implementation results show that the total cost of Montgomery multiplier is 130 KGates, the clock frequency is 180 MHz and the throughput of 1024-bit RSA encryption is 352 kbps. This design is suitable to be used in high speed RSA or ECC encryption/decryption. As a scalable design, it supports any key-length encryption/decryption up to the size of on-chip memory.

Color Demosaicing Using Asymmetric Directional Interpolation and Hue Vector Smoothing

TAKAHASHI, Y., HIRAKI, K., KIKUCHI, H., MURAMATSU, S. — 2008-04-14

This paper presents a color demosaicing method applied to the Bayer pattern color filter array (CFA). Reliable estimation of an edge direction, edge-directed asymmetric interpolation, and the use of color samples at immediate neighbors are considered as the key guidelines for smooth and sharp image restoration. Also, special interest is directed to local areas that are rich in high spatial frequency variations. For suppression of false colors likely to occur in those areas, a hue vector representation is introduced so that the spatial correlation between different color components may be exploited in consistent with the local constant-hue principle. Smoothing is repeated in the hue vector field a few times. Experimental results have shown preferable performances in terms of PSNR, CIELAB color difference, hue angle difference, CIE chromaticity and visual appearance, in particular resulting in less false colors.

Parallel Improved HDTV720p Targeted Propagate Partial SAD Architecture for Variable Block Size Motion Estimation in H.264/AVC

HUANG, Y., LIU, Z., SONG, Y., GOTO, S., IKENAGA, T. — 2008-04-14

One hardware efficient and high speed architecture for variable block size motion estimation (VBSME) in H.264 is presented in this paper. By improving the pipeline structure and processing element (PE) circuits, the system latency and hardware cost is reduced, which makes this structure more hardware efficient than the original Propagate Partial SAD architecture. For small and middle frame size picture's coding, the proposed structure can save 12.1% hardware cost compared with original Propagate Partial SAD structure. In the case of HDTV, since small inter modes trivially contribute to the coding quality, we remove modes below 8 x 8 in our design. By adopting mode reduction technique, when the set number of PE array is less than 8, the proposed mode reduction based Propagate Partial SAD structure can work at faster clock speed and consume less hardware cost than widely used SAD Tree architecture. It is more robust to the high speed timing constraint when parallel processing is considered. With TSMC 0.18µm technology in worst work conditions (1.62 V, 125°C), its peak throughput of 8-set PE array structure is 720p@30 Hz with 128 x 64 search range and 5 reference frames. 12 k gates hardware cost can be reduced by our design compared with the parallel SAD Tree architecture.

A New Progressive Image Quality Control Method for ROI Coding in JPEG2000 Standard

KITAURA, Y., MUNEYASU, M., NAKANISHI, K. — 2008-04-14

JPEG2000 still image coding standard has a feature called Region of Interest (ROI) coding. This feature can encode a restricted region in an image prior to than its background (BG) region. In low bit rate compression, the code of the ROI region occupies the most of the bit stream in the whole image and it causes the serious deterioration of the image quality in the BG region. This paper proposes a new image quality control method between the ROI region and the BG one by the one time encoding process and it can achieve more detailed image quality control. The use of ROI masks in the encoder makes it possible. The standard decoder of JPEG2000 part1 can decode the encoded data in the proposed method.

Functionally Layered Video Coding for Water Level Monitoring

UDOMSIRI, S., IWAHASHI, M., MURAMATSU, S. — 2008-04-14

This paper proposes a new type of layered video coding especially for the use of monitoring water level of a river. A sensor node of the system decomposes an input video signal into some kinds of component signals and produces a bit stream functionally separated into three layers. The first layer contains the minimum components effective for detecting the water level. It is transmitted at very low bit rate for regular monitoring. The second layer contains signals for thumb-nail video browsing. The third layer contains additional data for decoding the original video signal. These are transmitted in case of necessity. A video signal is decomposed into several bands with the three dimensional Haar transform. In this paper, optimum bands to be contained into the 1st layer are experimentally investigated considering both of water level detection and data size to be transmitted. As a result, bit rate for transmitting the first layer is reduced by 32.5% at the cost of negligible 3.7% decrease of recognition performance for one of video examples.

Adaptive Search Range Algorithms for Variable Block Size Motion Estimation in H.264/AVC

CHEN, Z., SONG, Y., IKENAGA, T., GOTO, S. — 2008-04-14

Comparing with search pattern motion estimation (ME) algorithms, adaptive search range (ASR) algorithms are more fundamental, regular and flexible. In variable block size motion estimation (VBSME), ASR algorithms can be applied whether on a whole frame (frame level), or on an entire macroblock which includes up to forty-one blocks (macroblock level), or just on a single block (block level). In the other hand, in H.264/AVC, not the motion vectors (MVs) but the motion vector differences (MVDs) are coded and the median motion vector predictors (median-MVPs) are used to place the search centers. In this sense, it can be thought that the search windows (SWs) are centered at the positions pointed by median-MVPs, the search ranges (SRs) play the role of limiting MVDs. Thus it is reasonable for considering using MVDs to predict SRs. In this paper, one of the MB level and two of the block level, at all three MVD based SR prediction algorithms are proposed. VBSME based experiments are carried out to assess the proposed algorithms. Comparisons between the proposed three algorithms and the previously proposed one given in [8] are done in terms of encoding quality and computational complexity.

A Palmprint Recognition Algorithm Using Phase-Only Correlation

ITO, K., AOKI, T., NAKAJIMA, H., KOBAYASHI, K., HIGUCHI, T. — 2008-04-14

This paper presents a palmprint recognition algorithm using Phase-Only Correlation (POC). The use of phase components in 2D (two-dimensional) discrete Fourier transforms of palmprint images makes it possible to achieve highly robust image registration and matching. In the proposed algorithm, POC is used to align scaling, rotation and translation between two palmprint images, and evaluate similarity between them. Experimental evaluation using a palmprint image database clearly demonstrates efficient matching performance of the proposed algorithm.

Cross-Correlation by Single-bit Signal Processing for Ultrasonic Distance Measurement

HIRATA, S., KUROSAWA, M. K., KATAGIRI, T. — 2008-04-14

Ultrasonic distance measurement using the pulse-echo method is based on the determination of the time of flight of ultrasonic waves. The pulse-compression technique, in which the cross-correlation function of a detected ultrasonic wave and a transmitted ultrasonic wave is obtained, is the conventional method used for improving the resolution of distance measurement. However, the calculation of a cross-correlation operation requires high-cost digital signal processing. This paper presents a new method of sensor signal processing within the pulse-compression technique using a delta-sigma modulated single-bit digital signal. The proposed sensor signal processing method consists of a cross-correlation operation employing single-bit signal processing and a smoothing operation involving a moving average filter. The proposed method reduces the calculation cost of the digital signal processing of the pulse-compression technique.

A Performance Optimized Architecture of Deblocking Filter in H.264/AVC

MIN, K.-Y., CHONG, J.-W. — 2008-04-14

In this paper, we propose memory and performance optimized architecture to accelerate the operation speed of adaptive deblocking filter for H.264/JVT/AVC video coding. The proposed deblocking filter executes loading/storing and filtering operations with only 192 cycles for 1 macroblock. Only 2x4x4 internal buffers and 32x16 internal SRAM are adopted for the buffering operation of deblocking filter with I/O bandwidth of 32 bit. The proposed architecture can process the filtering operation for 1 macroblock with less filtering cycles and lower memory sizes than some conventional approaches of realizing deblocking filter. The efficient hardware architecture is implemented with novel data arrangement, hybrid filter scheduling and minimum number of buffer. The proposed architecture is suitable for low cost and real-time applications, and the real-time decoding with 1080HD (1920x1088@30 fps) can be easily achieved when working frequency is 70 MHz.

Novel Register Sharing in Datapath for Structural Robustness against Delay Variation

INOUE, K., KANEKO, M., IWAGAKI, T. — 2008-04-14

As the feature size of VLSI becomes smaller, delay variations become a serious problem in VLSI. In this paper, we propose a novel class of robustness for a datapath against delay variations, which is named structural robustness against delay variation (SRV), and propose sufficient conditions for a datapath to have SRV. A resultant circuit designed under these conditions has a larger timing margin to delay variations than previous designs without sacrificing effective computation time. In addition, under any degree of delay variations, we can always find an available clock frequency for a datapath having SRV property to operate correctly, which could be a preferable characteristic in IP-based design.

Low Power LDPC Code Decoder Architecture Based on Intermediate Message Compression Technique

SHIMIZU, K., TOGAWA, N., IKENAGA, T., GOTO, S. — 2008-04-14

Reducing the power dissipation for LDPC code decoder is a major challenging task to apply it to the practical digital communication systems. In this paper, we propose a low power LDPC code decoder architecture based on an intermediate message-compression technique which features as follows: (i) An intermediate message compression technique enables the decoder to reduce the required memory capacity and write power dissipation. (ii) A clock gated shift register based intermediate message memory architecture enables the decoder to decompress the compressed messages in a single clock cycle while reducing the read power dissipation. The combination of the above two techniques enables the decoder to reduce the power dissipation while keeping the decoding throughput. The simulation results show that the proposed architecture improves the power efficiency up to 52% and 18% compared to that of the decoder based on the overlapped schedule and the rapid convergence schedule without the proposed techniques respectively.

A Novel Expression of Spatial Correlation by a Random Curved Surface Model and Its Application to LSI Design

OHKAWA, S.-i., MASUDA, H., INOUE, Y. — 2008-04-14

We have proposed a random curved surface model as a new mathematical concept which enables the expression of spatial correlation. The model gives us an appropriate methodology to deal with the systematic components of device variation in an LSI chip. The key idea of the model is the fitting of a polynomial to an array of Gaussian random numbers. The curved surface is expressed by a new extension from the Legendre polynomials to form two-dimensional formulas. The formulas were proven to be suitable to express the spatial correlation with reasonable computational complexity. In this paper, we show that this approach is useful in analyzing characteristics of device variation of actual chips by using experimental data.

Fujimaki-Takahashi Squeeze: Linear Time Construction of Constraint Graphs of Floorplan for a Given Permutation

TAKAHASHI, T., FUJIMAKI, R. — 2008-04-14

A floorplan is a subdivision of a rectangle into rectangular faces with horizontal and vertical line segments. We call a floorplan room-to-room when adjacencies between rooms are considered. Fujimaki and Takahashi showed that any room-to-room floorplan can be represented as a permutation. In this paper, we give an O(n)-time algorithm that constructs the vertical and the horizontal constraint graphs of a floorplan for a given permutation under this representation.

Prevention in a Chip of EMI Noise Caused by X'tal Oscillator

KUROKAWA, A., FUJITA, H., IBE, T. — 2008-04-14

Developing LSIs with EMI suppression, particularly for use in automobiles, is important for improving warranties and customer acquisition. First, we describe that the measures against EMI noise caused by a X'tal oscillator are important. Next, we present a practical method for analyzing the noise with models of the inside and outside of a chip. In addition, we propose a within-chip measure against EMI noise that takes chip cost into account. The noise is suppressed by using an appropriate resistance and capacitance on the power line. Simulation results demonstrated the method's effectiveness in suppressing noise.

A Behavioral Synthesis Method with Special Functional Units

SADAKATA, T., MATSUNAGA, Y. — 2008-04-14

This paper proposes a novel Behavioral Synthesis method that tries to reduce the number of clock cycles under clock cycle time and total functional unit area constraints using special functional units efficiently. Special functional units are designed to have shorter delay and/or smaller area than the cascaded basic functional units for specific operation patterns. For example, a Multiply-Accumulator is one of them. However, special functional units may have less flexibility for resource sharing because intermediate operation results may not be able to be obtained. Hence, almost all conventional methods can not handle special functional units efficiently for the reduction of clock cycles in practical time, especially under a tight area constraint. The proposed method makes it possible to solve module selection, scheduling, and functional unit allocation problems using special functional units in practical time with some heuristics. Experimental results show that the proposed method has achieved maximally 33% reduction of the cycles for a small application and 14% reduction for a realistic application in practical time.

Issue Mechanism for Embedded Simultaneous Multithreading Processor

ZANG, C., IMAI, S., FRANK, S., KIMURA, S. — 2008-04-14

Simultaneous Multithreading (SMT) technology enhances instruction throughput by issuing multiple instructions from multiple threads within one clock cycle. For in-order pipeline to each thread, SMT processors can provide large number of issued instructions close to or surpass than using out-of-order pipeline. In this work, we show an efficient issue logic for predicated instruction sequence with the parallel flag in each instruction, where the predicate register based issue control is adopted and the continuous instructions with the parallel flag of '0' are executed in parallel. The flag is pre-defined by a compiler. Instructions from different threads are issued based on the round-robin order. We also introduce an Instruction Queue skip mechanism for thread if the queue is empty. Using this kind of issue logic, we designed a 6 threads, 7-stage, in-order pipeline processor. Based on this processor, we compare round-robin issue policy (RR(T₁–T_n)) with other policies: thread one always has the highest priority (PR(T₁)) and thread one or thread n has the highest priority in turn (PR(T₁–T_n)). The results show that RR(T₁–T_n) policy outperforms others and PR(T₁–T_n) is almost the same to RR(T₁–T_n) from the point of view of the issued instructions per cycle.