Earticle

The authors propose the side information-free low-computational complexity Time Domain Sequence Superposition - Selected Mapping (TDSS-SLM) scheme for PAPR suppression. We adopt IFFT point reduction for transmitter and reception technique using pilot subcarriers for side information estimation. The simulation results show the [R1-1]TDSS-SLM transmitter achieves PAPR reduction of 2.3 dB which corresponds to the 1.5 times larger PAE than that in the conventional transmitter. In addition, the proposed receiver shows loss-less BER performances for both AWGN channel and two types of fading channels, while the computational complexity of the proposed receiver is nearly the same as the conventional OFDM receiver, which is less than 1/10 of the ML receiver.

Since most studies for estimating an angle-of-arrival (AOA) based on the antenna array have considered the antenna array with a single configuration, they are not proper to simultaneously estimate AOAs of multiple signals with various frequencies. In this paper, we introduce a cascade AOA estimation technique consisting of CAPON and Beamspace Multiple Signal Classification (MUSIC), based on a Combined Array Antenna (CAA) with Uniform Rectangular Frame Array (URFA) and Uniform Circular Array (UCA), for enhancing the above problem. In addition, we provide the computational complexity analysis for showing the low computational complexity of this technique comparing to the conventional technique.

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, We know the proposed algorithm is prefer than the existent algorithm.

본 논문은 계산량이 인 새로운 형태의 QRD-LS 적응 알고리즘을 제안한다. 제안한 알고리즘의 주요 사항은 입력벡터의 단위벡터 계산이 Given Rotation 과정에서 이루어짐에 근거하고 있다. 알고리즘의 성능 평가는 FIR 시스템 식별 문제를 컴퓨터 시뮬레이션을 통하여 수행하였다. 이 알고리즘은 시뮬레이션의 결과 좋은 성능을 나타내었다. 그리고 이론적으로 평균 측면에서 알고리즘이 최적 계수 벡터에 수렴함을 보였다

This paper proposes a new QRD-LS adaptive algorithm with computational complexity of . The main idea of proposed algorithm(D-QR-RLS) is based on the fact that the computation for the unit vector of is made from the process during Givens Rotation. The performance of the algorithm is evaluated through computer simulation of FIR system identification problem. As verified by simulation results, this algorithm exhibits a good performance. And, we can see the proposed algorithm converges to optimal coefficient vector theoretically.

With the rapid development of social network, more and more information is being presented in the form of multimedia, so the security of multimedia files, especially video files, has been capturing the attention of the researchers. Video encryption is widely applied to the DRM system, video conference, medical and military fields and so on. A general video encryption algorithm is proposed to solve the problem that the existing multiple formats (compression formats and container formats) of video requires a unified encryption scheme. Meanwhile, the proposed scheme can not only provide variable levels of security, but also have good performance of compression efficiency and computational complexity, which can ensure that the encrypted video data can be operation.

We consider a problem of the selection and scheduling of movies for a multiplex (i.e., a theater with multiple screens) to maximize the exhibitor's cumulative revenue over a fixed planning horizon. Effective and timely decisions on screen management directly and crucially relate to the effective capacity and resources usage, productivity and successful economical performance of the theater. In the problem under study, the release times of the movies that can potentially be selected during the planning horizon are known a priori. If selected for screening, a movie must be played continuously through its obligatory duration, after which its run may or may not be extended. The problem involves two primary decisions: (i) the selection of movies to play and (ii) the determination of the duration of screening for the selected movies. We show that revenue optimization problem under the preempt-resume policy is strongly NP-hard even in the minimal case, thus completely resolving the complexity status of the problem. We also consider the optimization problem under the non-preempt policy for a case of unequal screen capacities and show that an optimal solution may significantly differ from the solution obtained for the equal-capacity case.

This paper explores the contributions of all reconstruction combinations of the synthesis delay filters in adaptive synthesis filter banks and their effects on objective performance in image compression. It presents a novel comprehensive approach to manage all reconstruction combinations and their placing configuration in the adaptive synthesis filter banks. Based on the contributions of reconstruction combinations, it suggests the most suitable reconstruction combinations in order to reduce the computation complexity of adaptive synthesis filter banks. The simulations are carried out by using popular bi-orthogonal 9/7 filters on a number of standard test images with different design examples, which show that suggested reconstruction combinations of the synthesis filters play a very vital role on objective performance.

다중 규격, 다중 대역, 다중 서비스 시스템간의 유연한 인터페이스를 보장하기 위한 SDR(Software Defined Radio)기술의 구현을 위해서는 안정성 및 저전력, 저연산량의 DDC(Digital Down Conversion)기술이 필수적이다. DDC기술은 디지털 채널 필터로 이루어진다. 이 때 일반적인 디지털 필터는 유한어장으로 인하여 오버플로우나 반올림 오차에 취약한 단점이 있다. 이에 본 논문에서는 이러한 단점을 보완하는 DDC 구조를 제안하였다. 그 방법으로 WDF(Wave Digital Filter)를 이용한 구조는 그 구조상 반올림 오차에 의한 잡음에 강하다. 따라서 필터계수의 단어길이가 짧을 경우 유용하게 사용된다. 또한 IIR기반의 필터이기 때문에 FIR기반의 필터보다 탭수가 줄어들므로 연산량이 줄어든다. 제안한 DDC구조에 사용된 CIC(Cascaded Integrator Comb) 필터, WDF, IFOP(Interpolated Fourth-Order Polynomials)에 대하여 분석하였으며, 모의실험을 통하여 결과를 확인하였다.

Multi-standard, multi-band, multi-service system to ensure a flexible interface between the SDR (Software Defined Radio) technology for the implementation of the Stability and Low-Power, Low-Calcualrion DDC (Digital Down Conversion) technology is essential. DDC technology consists of a digital channel filter. This is a typical digital filter because of the limited fisheries are vulnerable to overflow and rounding errors are drawbacks. In this paper, we overcome this disadvantage, we propose the structure of the DDC. The way WDF (Wave Digital Filter) Structural rounding error due to the structural resistance to noise. Therefore, This is the useful structure when the filter coefficients's word length is short. In addition, since IIR filters based on FIR filters based on the amount of computation is reduced because fewer than filter's tap. The proposed structure is used in DDC that CIC (Cascaded Integrator Comb) filter, WDF, IFOP (Interpolated Fourth-Order Polynomials) were analyzed with respect to, the results were confirmed by computer simulation.

In the next generation wireless communication system, the beamforming technique based on a massive antenna is one of core technologies for transmitting and receiving huge amounts of data, efficiently and accurately. For highly performed and highly reliable beamforming, it is required to accurately estimate the Angle of Arrival (AOA) for the desired signal incident to an antenna. Employing the massive antenna with a large number of elements, although the accuracy of the AOA estimation is enhanced, its computational complexity is dramatically increased so much that real-time communication is difficult. In order to improve this problem, AOA estimation algorithms based on the massive antenna with the low computational complexity have been actively studied. In this paper, we compute and analyze the computational complexity of the cascade AOA estimation algorithm based on the Flexible Massive Concentric Circular Array (FMCCA). In addition, its computational complexity is compared to conventional AOA estimation techniques such as the Multiple Signal Classification (MUSIC) algorithm with the high resolution and the Only Beamspace MUSIC (OBM) algorithm.

The objective of the paper is to compare the computational complexity of five algorithms for computing time-domain second-order Volterra filter outputs in terms of number of real multiplication and addition operations required for implementation. This study shows that if the filter memory length is greater that or equal to 16, the fast algorithm using the overlap-save method and the frequency-domain symmetry properties of the quadratic coefficients is the most efficient among the algorithms investigated in this paper, When the filter memory length is less than 16, the algorithm using the time-domain symmetry properties is better than any other algorithm.

This paper investigates the relationship between the complexity of a phonological pattern and its learnability. The Complexity Hypothesis refers to the idea that less complex patterns are easier to learn than more complex patterns. In this paper, the hypothesis is tested as the Subregular Hypothesis, which measures computational complexity in terms of the Subregular Hierarchy. This hypothesis states that patterns in a lower class are less complex and thus easier to learn. Two artificial grammars of vowel harmony patterns with neutral vowels were tested. The two patterns differed in terms of the computational complexity and attestedness. The At Least One (ALO) pattern, the less complex pattern and the unattested vowel harmony pattern, is predicted to be easier to learn than the Rightmost pattern, the more complex pattern and the attested pattern. The results showed that the Rightmost pattern was more difficult to learn than the ALO pattern. This supports the Subregular Hypothesis and implies a learning bias toward computational complexity despite the attestedness of vowel harmony patterns.

In this paper, we propose a modified cubic convolution interpolation for the enlargement or reduction of digital images using a pixel difference value. The proposed method has a low complexity: the number of multiplier of weighted value to calculate one pixel of a scaled image has seven less than that of cubic convolution interpolation has sixteen. We use the linear function of the cubic convolution and the difference pixel value for selecting interpolation methods. The proposed method is compared with the conventional one for the computational complexity and the image quality. The simulation results show that the proposed method has less computational complexity than one of the cubic convolution interpolation.

In this paper, we propose a new method for reducing the number of distance calculations in the LBG (Linde, Buzo, Gray) algorithm, which is widely used method to construct a codebook in vector quantization of speech recognition system. The proposed algorithm can reduce the distance calculation between input vector and codeword by utilizing the observation that codewords are quickly stabilized as the number of iteration increases. From the simulation results, it is shown that we can reduce the running times over $43.77\%$ on average in comparison with current LBG algorithm without sacrificing the performance of codebook.

A technique for realizing linear phase IIR filters has been proposed by Powell-Chau which gives a real-time implementation of H(z-1).H(z), where H(z) is a causal nonlinear phase IIR filter. Powell-Chau system is linear but not timeinvariant system. Therefore, that system has group delay response that exhibits a minor sinusoidal variation superimposed on a constant value. In the signal processing, this oscillation seriously degrade the signal quality. Unfortunately, that system has a large sample delay of 4L and also more computational complexity. Proposed system is present a reduced computational complexity technique by moved the numerator polynomial of H(1/z) out to cascade with causal filter H(z) and remain only all-pole of H(1/z), then applied truncated infinite impulse response to finite with truncated IIR filtel $H_L$(z) and L sample delay to subtract the output sequence from the top and bottom filter. Proposed system is linear time invariance and group delay response and total harmonic distortion are also improved.

This study investigated the effect of sentiment in second language (L2) writing prompts on the syntactic complexity and emotionality in essays written by English as a Foreign Language (EFL) learners. The research analyzed a dataset of 1,004 essays written by Korean adolescent students to examine the relationship between the sentiment scores of writing prompts and learner essays. The study used computational tools to measure various dimensions of syntactic complexity. The findings indicated that the sentiment of writing prompts is a significant predictor of the emotionality scores of L2 students' essays. The positive correlation suggested that more positive prompts are more likely to elicit overall more positive essays. Additionally, the study identified significant differences in baseline sentiment values between writing prompts and learner essays. Even when responding to highly negative prompts, L2 students tended to use neutral sentiment in their essays. This study demonstrated how emotional prompts can impact cognitive load, resulting in simpler syntactic structures. It also highlighted the potential risks in educational assessment due to construct-irrelevant variance. Among the indices used to measure syntactic complexity, mean length of clauses (MLC) and coordinate phrases per clause (CP/C) showed significant effects influenced by the emotionality of writing prompts and learner essays. The implications of this study are discussed.

대규모 배열 안테나 기반 통신 시스템에서 수신신호에 포함된 신호의 개수를 정확하게 추정하는 것은 빔형성 및 도래각 추정 성능을 결정하는 중요한 요소이다. 그러나, 기존 요소공간 기반 신호개수 추정 방식은 배열 요소 수가 증가할수록 공분산 행렬 계산과 기준식 계산에 필요한 연산량이 급격히 증가하여 대규모 배열 안테나 환경에서 실시간 신호처리 구현을 어렵게 만드는 한계가 있다. 이러한 문제를 해결하기 위해 복잡도 감소를 위한 다양한 연구가 이루어져 왔는데, 본 논문은 정방배열 안테나에 대한 공분산 행렬 계산, 고유값 분해, AIC 및 MDL 기준식 계산 과정에서 요구되는 연산량을 기반으로 한 정량적 복잡도 모델을 제시한다. 또한, 컴퓨터 시뮬레이션을 통해 빔공간 변환을 기반으로 차원을 축소한 빔공간 기반 접근법이 요소공간 대비 계산 복잡도를 효과적으로 줄일 수 있음을 확인하였다.

In large-scale array antenna-based communication systems, accurately estimating the number of signals contained in the received signal is essential for achieving reliable beamforming and AOA(: Angle-of-Arrival) estimation performance. However, conventional element-space-based signal number estimation methods exhibit significant limitations, as the computational burden for covariance matrix computation and criterion evaluation increases rapidly with the number of array elements, making real-time processing difficult in large-scale scenarios. In order to address this issue, various studies have investigated methods for reducing computational complexity. In this paper, we provide a quantitative complexity model for an uniform rectangular array antenna based on operations required for covariance computation, eigenvalue decomposition, and AIC/MDL criterion evaluation. Furthermore, computer simulation results demonstrate that the beamspace-based approach, which reduces dimensionality via beamspace transformation, can effectively decrease computational complexity compared to the element-space method.

MoM은 대표적인 적분방정식기반 full-wave simulation 방법이며, 이는 MLFMM 방법을 적용하여 효율적으로 계산될 수 있다. MoM 또는 MLFMM 방법에서 대규모 산란체 표면전류를 계산하는 과정에는 주로 반복법들이 사용된다. 이 가운데 BiCGstab(l)은 l값이 증가할수록 반복횟수는 줄어들지만, 반복당 수행되는 연산횟수가 증가하는 특징이 있다. 본 논문에서는 MLFMM 방법에 적용된 BiCGstab(l) 반복법의 l값에 따른 수렴속도와 연산량을 분석한 후, 효율적인 l값을 제안한다.

The method of moments(MoM) is one of the most popular integral-equation-based full-wave simulation methods, and the multi-level fast multipole method(MLFMM) algorithm can be used for its efficient calculation. When calculating the surface current on the large scatterer in the MoM or MLFMM, iterative methods for the final matrix inversion are used. Among them, BiCGstab(l) has been widely adopted due to its good convergence rate. The number of iterations can be reduced when l becomes larger, but the number of operations per iteration is increased. Herein, we analyze the computational complexity of BiCGstab(l) in the MLFMM method and propose an optimum choice of l.

Since computing the log-likelihood of Gaussian mixture density is a major computational burden for the speech recognizer based on the continuous HMM, several techniques have been proposed to reduce the number of mixtures to be used for recognition. In this paper, we propose a modified Bucket Box Intersection (BBI) algorithm, in which two relative thresholds are employed: one is the relative threshold in the conventional BBI algorithm and the other is used to reduce the number of the Gaussian boxes which are intersected by the hyperplanes at the boxes' edges. The experimental results show that the proposed algorithm reduces the number of Gaussian mixtures by 12.92% during the recognition phase with negligible performance degradation compared to the conventional BBI algorithm.

본 논문에서는 영상에서 반복적인 가우시안 커널을 사용하여 소형 표적의 크기와 위치를 검출하는 방법을 제안한다. 소형 표적의 크기와 위치를 검출하는 방법은 우선 페이싯 모델을 원 영상에 적용하여 국부 극대 값을 검출하고 3x3 가우시안 커널을 반복적으로 사용한다. 이때 반복횟수에 따른 국부 극대값 를 비교하여 이에 따른 소형 표적의 크기를 결정한다. 또한 계산의 복잡성을 줄이기 위하여 커널을 반복적으로 사용할 때 가우시안 피라미드를 사용하였다. 실험에서는 소형 표적의 크기와 위치가 반복 횟수에 따라 정확히 검출되는 것을 확인하였고 기존의 방법에 비하여 처리속도가 개선된 것을 확인하였다.

In this paper, we propose a technique to detect the size and location of the small target in images by using Gaussian kernel repeatedly. In order to detect the size and location of the small target, we find the local maximum value by applying the facet model and then use the 3x3 Gaussian kernel repeatedly. we determine the size of small target by comparing the local maximum value according to the number of iteration. To reduce the computational complexity, we use the Gaussian pyramid when using the kernel repeatedly. Through the experiment, we verified that the size and location of the small target is detected by the number of iterations and results show improvements from conventional methods.