Contourlet transform (CT) is a new image representation method, which can efficiently represent contours and textures in images. However, CT is a kind of overcomplete transform with a redundancy factor of 4/3. If it is applied to image compression straightforwardly, the encoding bit-rate may increase to meet a given distortion. This fact baffles the coding community to develop CT-based image compression techniques with satisfactory performance. In this paper, we analyze the distribution of significant contourlet coefficients in different subbands and propose a new contourlet-based embedded image coding (CEIC) scheme on low bit-rate. The well-known wavelet-based embedded image coding (WEIC) algorithms such as EZW, SPIHT and SPECK can be easily integrated into the proposed scheme by constructing a virtual low frequency subband, modifying the coding framework of WEIC algorithms according to the structure of contourlet coefficients, and adopting a high-efficiency significant coefficient scanning scheme for CEIC scheme. The proposed CEIC scheme can provide an embedded bit-stream, which is desirable in heterogeneous networks. Our experiments demonstrate that the proposed scheme can achieve the better compression performance on low bit-rate. Furthermore, thanks to the contourlet adopted in the proposed scheme, more contours and textures in the coded images are preserved to ensure the superior subjective quality.

This paper is about an efficient implementation of adaptive filtering for echo cancelers. The first objective of this paper is to propose a simplified method of the flexible block Multi-Delay Filter (MDF) algorithm in the time-domain. Then, we will derive a new method for the step-size adaptation coefficient. The second objective is about the realization of a Block Proportionate Normalized Least Mean Squares (BPNLMS++) with the simplified MDF (SMDF) implementation. Using the new step-size method and the smaller block dimension proposed by SMDF, we achieve a faster convergence of the adaptive process with a limited computational cost. Then, an efficient implementation of the new procedure (SMDF-BPNLMS++) block filtering is proposed using Fermat Number Transform, which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor.

In this letter, we propose a simple detection ordering which can be applied to successive interference cancellation in low-complexity equalization of OFDM in doubly selective channels. Since the detection ordering can be determined solely on the basis of pilot symbol arrangement, the proposed scheme can lead to a significant performance improvement without additional complexity on the premise that pilot symbols are employed primarily for channel estimation.

In this paper, a new fractional sample rate conversion (SRC) scheme based on a direct insertion/cancellation scheme is proposed. This scheme is suitable for signals that are sampled at a high sample rate and converted to a lower sample rate. The direct insertion/cancellation scheme may achieve low-complexity and lower power consumption as compared to the other SRC techniques. However, the direct insertion/cancellation technique suffers from large aliasing and distortion. The aliasing from an adjacent channel interferes the desired signal and degrades the performance. Therefore, a modified direct insertion/cancellation scheme is proposed in order to realize high performance resampling.

In mobile communication systems, high speed packet data services are demanded. In the high speed data transmission, throughput degrades severely due to severe inter-path interference (IPI). Recently, we proposed a random transmit power control (TPC) to increase the uplink throughput of DS-CDMA packet mobile communications. In this paper, we apply IPI cancellation in addition to the random TPC. We derive the numerical expression of the received signal-to-interference plus noise power ratio (SINR) and introduce IPI cancellation factor. We also derive the numerical expression of system throughput when IPI is cancelled ideally to compare with the Monte Carlo numerically evaluated system throughput. Then we evaluate, by Monte-Carlo numerical computation method, the combined effect of random TPC and IPI cancellation on the uplink throughput of DS-CDMA packet mobile communications.

In the Intelligent Transportation System, millimeter waves are used and antennas are required beam scanning ability. In the millimeter wave operation, a lens antenna is one of the prominent candidates which achieves wide angle beam scanning. Wide angle scanning can be achieved by introducing Abbe sine condition to lens surface shaping. Authors designed the shaped lens antenna that could achieve beam scanning 30. The narrow beam widths were maintained on the scanning plane. However, the beam widths were broadened on the transverse plane and large gain reduction was appeared. It was clarified that the reason of this beam deterioration was due to the phase delay on the antenna aperture. In this paper, an array feed composed of a group of rectangular horns is employed to compensate the phase delay on the antenna aperture. In designing the array feed, because there were no examples of phase radiation pattern synthesis, a new radiation pattern synthesis method is studied. Ability of the weighting matrix contained in the Least Mean Square synthesis method is paid attention. Adequate weighting matrix is found out. Satisfactory phase radiation pattern that can compensate the phase delay and an adequate amplitude radiation pattern are achieved. As a result, the improvement of scanned beam widths and antenna gains through the array feed are ensured. And adequate horn arrangements of the array feed for improving scanned beam are clarified. Moreover, in order to examine the realization of an actual array feed, the exact electromagnetic simulation is conducted. The validity of the radiation pattern synthesis is clarified.

This paper presents a triple-band WCDMA direct conversion receiver (DCR) IC that needs a small number of off-chip components and control signals from digital baseband (DBB) IC. The DCR IC consists of 3 quadrature demodulators (QDEMs) with on-chip impedance matching circuit and an analog baseband block (ABB) that contains a low-pass filter (LPF) with fc automatic tuning circuit using no off-chip components and a linear-in-dB variable-gain amplifier (VGA) with on-chip analog high-pass filter (HPF). In order to make use of DBB control-free DC offset canceler, the DCR is designed to avoid large gain change under large interference that causes long transient response. In order to realize that characteristic without increasing quiescent current, the QDEM is used that employs class AB input stage and low-noise common mode feedback (CMFB) output stage. The DCR IC was fabricated in a SiGe BiCMOS process and occupies about 2.9 mm3.0 mm. The DCR needs SAW filters only for off-chip components and a gain control signal from DBB IC for AGC loop. The IIP3 of over -4.4 dBm for small signal input level and that of over +1.9 dBm for large signal input level are achieved. The gain compression of the desired signal is less than 0.3 dB for ACS Case-II condition.

The performance of the adaptive scan for spaceborne rain radar, which uses a quick scan for rain search followed by a normal or concentrated scan only for rainy areas, are studied through a simulation using TRMM (Tropical Rainfall Measuring Mission) Precipitation Radar (PR) data. Trade-off studies are performed to find an optimum quick-scan and rain search method to minimize rain missing and false alarm of rain area. Using the optimum method thus determined, consecutive 8-day TRMM PR data are used to statistically evaluate the performance of the adaptive scan in terms of sensitivity improvement and power consumption saving. It is shown that more than 3-dB improvement in effective signal-to-noise ratio (SNe) can be achieved for 40% of the total observations. Alternatively, about 26% power saving can be achieved if the SNe is kept the same.

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.

Recently, a new frequency-domain equalization (FDE) technique, called overlap FDE, that requires no GI insertion was proposed. However, the residual inter/intra-block interference (IBI) cannot completely be removed. In addition to this, for multicode direct sequence code division multiple access (DS-CDMA), the presence of residual inter-chip interference (ICI) after FDE distorts orthogonality among the spreading codes. In this paper, we propose an iterative overlap FDE for multicode DS-CDMA to suppress both the residual IBI and the residual ICI. In the iterative overlap FDE, joint minimum mean square error (MMSE)-FDE and ICI cancellation is repeated a sufficient number of times. The bit error rate (BER) performance with the iterative overlap FDE is evaluated by computer simulation.

Abnormal traffic that causes various problems on the Internet, such as P2P flows, DDoS attacks, and Internet worms, is increasing; therefore, the importance of methods that identify and control abnormal traffic is also increasing. Though the application of frequent-itemset-mining techniques is a promising way to analyze Internet traffic, the huge amount of data on the Internet prevents such techniques from being effective. To overcome this problem, we have developed a simple frequent-itemset-mining method that uses only a small amount of memory but is effective even with the large volumes of data associated with broadband Internet traffic. Using our method also involves analyzing the number of distinct elements in the itemsets found, which helps identify abnormal traffic. We used a cache-based implementation of our method to analyze actual data on the Internet and demonstrated that such an implementation can be used to provide on-line analysis of data while using only a small amount of memory.

A tree-shellable function is a positive Boolean function which can be represented by a binary decision tree whose number of paths from the root to a leaf labeled 1 equals the number of prime implicants. In this paper, we consider the tree-shellability of DNFs with restrictions. We show that, for read-k DNFs, the number of terms in a tree-shellable function is at most k2. We also show that, for k-DNFs, recognition of ordered tree-shellable functions is NP-complete for k=4 and tree-shellable functions can be recognized in polynomial time for constant k.

This paper proposes the minimum transition random X-filling (MTR-fill) technique, which is a new X-filling method, to reduce the amount of power dissipation during scan-based testing. In order to model the amount of power dissipated during scan load/unload cycles, the total weighted transition metric (TWTM) is introduced, which is calculated by the sum of the weighted transitions in a scan-load of a test pattern and a scan-unload of a test response. The proposed MTR-fill is implemented by simulated annealing method. During the annealing process, the TWTM of a pair of test patterns and test responses are minimized. Simultaneously, the MTR-fill attempts to increase the randomness of test patterns in order to reduce the number of test patterns needed to achieve adequate fault coverage. The effectiveness of the proposed technique is shown through experiments for ISCAS'89 benchmark circuits.

In the acoustic echo canceller (AEC), the step-size parameter of the adaptive filter must be varied according to the situation if double talk occurs and/or the echo path changes. We propose an AEC that uses a sub-adaptive filter. The proposed AEC can control the step-size parameter according to the situation. Moreover, it offers superior convergence compared to the conventional AEC even when the double talk and the echo path change occur simultaneously. Simulations demonstrate that the proposed AEC can achieve higher ERLE and faster convergence than the conventional AEC. The computational complexity of the proposed AEC can be reduced by reducing the number of taps of the sub-adaptive filter.

Feature transformation in automatic text classification (ATC) can lead to better classification performance. Furthermore dimensionality reduction is important in ATC. Hence, feature transformation and dimensionality reduction are performed to obtain lower computational costs with improved classification performance. However, feature transformation and dimension reduction techniques have been conventionally considered in isolation. In such cases classification performance can be lower than when integrated. Therefore, we propose an integrated feature analysis approach which improves the classification performance at lower dimensionality. Moreover, we propose a multiple feature integration technique which also improves classification effectiveness.

In order to estimate fast time-varying channels exactly, the Inter-Carrier Interference (ICI) caused by time-varying fading channels in Orthogonal Frequency Division Multiplexing (OFDM) systems is analyzed based on the Basis Expansion Model (BEM). A channel estimation and ICI cancellation algorithm with low complexity is proposed. A special pilot sequence is designed to minimize the cost of computing the channel state information in the proposed algorithm. Based on the property of channel frequency impulse matrix, the ICI can be canceled iteratively in frequency domain. The complexity of the algorithm is analyzed theoretically. Through simulation, the algorithm is shown to be effective in estimating channel state information and in cancelling ICI.

In this paper, we propose a scheme for test data reduction which uses broadcaster along with bit-flipping circuit. The proposed scheme can reduce test data without degrading the fault coverage of ATPG, and without requiring or modifying the arrangement of CUT. We theoretically analyze the test data size by the proposed scheme. The numerical examples obtained by the analysis and experimental results show that our scheme can effectively reduce test data if the care-bit rate is not so much low according to the number of scan chains. We also discuss the hybrid scheme of random-pattern-based flipping and single-input-based flipping.

We developed test data compression scheme for scan-based BIST, aiming to compress test stimuli and responses by more than 100 times. As scan-BIST architecture, we adopt BIST-Aided Scan Test (BAST), and combines four techniques: the invert-and-shift operation, run-length compression, scan address partitioning, and LFSR pre-shifting. Our scheme achieved a 100x compression rate in environments where Xs do not occur without reducing the fault coverage of the original ATPG vectors. Furthermore, we enhanced the masking logic to reduce data for X-masking so that test data is still compressed to 1/100 in a practical environment where Xs occur. We applied our scheme to five real VLSI chips, and the technique compressed the test data by 100x for scan-based BIST.

In this paper, we presented a Design-for-Secure-Test (DFST) technique for pipelined AES to guarantee both the security and the test quality during testing. Unlike previous works, the proposed method can keep all the secrets inside and provide high test quality and fault diagnosis ability as well. Furthermore, the proposed DFST technique can significantly reduce test application time, test data volume, and test generation effort as additional benefits.

We propose a noise suppression algorithm with the Kalman filter theory. The algorithm aims to achieve robust noise suppression for the additive white and colored disturbance from the canonical state space models with (i) a state equation composed of the speech signal and (ii) an observation equation composed of the speech signal and additive noise. The remarkable features of the proposed algorithm are (1) applied to adaptive white and colored noises where the additive colored noise uses babble noise, (2) realization of high performance noise suppression without sacrificing high quality of the speech signal despite simple noise suppression using only the Kalman filter algorithm, while many conventional methods based on the Kalman filter theory usually perform the noise suppression using the parameter estimation algorithm of AR (auto-regressive) system and the Kalman filter algorithm. We show the effectiveness of the proposed method, which utilizes the Kalman filter theory for the proposed canonical state space model with the colored driving source, using numerical results and subjective evaluation results.