This paper proposes a high-quality computed tomography (CT) image reconstruction method from low-dose X-ray projection data. A state-of-the-art method, proposed by Xu et al., exploits dictionary learning for image patches. This method generates an overcomplete dictionary from patches of standard-dose CT images and reconstructs low-dose CT images by minimizing the sum of a data fidelity and a regularization term based on sparse representations with the dictionary. However, this method does not take characteristics of each patch, such as textures or edges, into account. In this paper, we propose to classify all patches into several classes and utilize an individual dictionary with an individual regularization parameter for each class. Furthermore, for fast computation, we introduce the orthogonality to column vectors of each dictionary. Since similar patches are collected in the same cluster, accuracy degradation by the orthogonality hardly occurs. Our simulations show that the proposed method outperforms the state-of-the-art in terms of both accuracy and speed.

This paper addresses pilot contamination in massive multiple-input multiple-output (MIMO) uplink. Pilot contamination is caused by reuse of identical pilot sequences in adjacent cells. To solve pilot contamination, the base station utilizes differences between the transmission frames of different users, which are detected via joint channel and data estimation. The joint estimation is regarded as a bilinear inference problem in compressed sensing. Expectation propagation (EP) is used to propose an iterative channel and data estimation algorithm. Initial channel estimates are attained via time-shifted pilots without exploiting information about large scale fading. The proposed EP modifies two points in conventional bilinear adaptive vector approximate message-passing (BAd-VAMP). One is that EP utilizes data estimates after soft decision in the channel estimation while BAd-VAMP uses them before soft decision. The other point is that EP can utilize the prior distribution of the channel matrix while BAd-VAMP cannot in principle. Numerical simulations show that EP converges much faster than BAd-VAMP in spatially correlated MIMO, in which approximate message-passing fails to converge toward the same fixed-point as EP and BAd-VAMP.

In the user-congested high-frequency band, radio frequency interference (RFI) is a dominant factor that degrades the detection performance of high-frequency surface wave radar (HFSWR). Up to now, various RFI suppression algorithms have been proposed while they are usually inapplicable to the compact HFSWR because of the minimal array aperture. Therefore, this letter proposes a novel RFI mitigation scheme for compact HFSWR, even for single antenna. The scheme utilized the robust principal component analysis to separate RFI and target, based on the time-frequency distribution characteristics of the RFI. The effectiveness of this scheme is demonstrated by the measured data, which can effectively suppress RFI without losing target signal.

Sparse modeling is one of the most active research areas in engineering and science. The technique provides solutions from far fewer samples exploiting sparsity, that is, the majority of the data are zero. This paper reviews sparse modeling in radio techniques. The first half of this paper introduces direction-of-arrival (DOA) estimation from signals received by multiple antennas. The estimation is carried out using compressed sensing, an effective tool for the sparse modeling, which produces solutions to an underdetermined linear system with a sparse regularization term. The DOA estimation performance is compared among three compressed sensing algorithms. The second half reviews channel state information (CSI) acquisitions in multiple-input multiple-output (MIMO) systems. In time-varying environments, CSI estimated with pilot symbols may be outdated at the actual transmission time. We describe CSI prediction based on sparse DOA estimation, and show excellent precoding performance when using the CSI prediction. The other topic in the second half is sparse Bayesian learning (SBL)-based channel estimation. A base station (BS) has many antennas in a massive MIMO system. A major obstacle for using the massive MIMO system in frequency-division duplex mode is an overhead for downlink CSI acquisition because we need to send many pilot symbols from the BS and to get the feedback from user equipment. An SBL-based channel estimation method can mitigate this issue. In this paper, we describe the outline of the method, and show that the technique can reduce the downlink pilot symbols.

The PCHS (Park-Chang-Hong-Seo) algorithm is a varied Karatsuba algorithm (KA) that utilizes a different splitting strategy with no overlap module. Such an algorithm has been applied to develop efficient hybrid GF(2m) multipliers for irreducible trinomials and pentanomials. However, compared with KA-based hybrid multipliers, these multipliers usually match space complexity but require more gates delay. In this paper, we proposed a new design of hybrid multiplier using PCHS algorithm for irreducible all-one polynomial. The proposed scheme skillfully utilizes redundant representation to combine and simplify the subexpressions computation, which result in a significant speedup of the implementation. As a main contribution, the proposed multiplier has exactly the same space and time complexities compared with the KA-based scheme. It is the first time to show that different splitting strategy for KA also can develop the same efficient multiplier.

Orthogonal frequency division multiplexing (OFDM) technique has been widely used in communication systems in pursuit of the most efficient utilization of spectrum. However, the increase of the number of orthogonal subcarriers will lead to the rise of the peak-to-average power ratio (PAPR) of the waveform, thus reducing the efficiency of the power amplifiers. In this letter we propose a phase-changed PAPR reduction technique based on windowing function architecture for OFDM systems. This technique is based on the idea of phase change, which makes the spectrum of output signal almost free of regrowth caused by peak clipping. It can reduce more than 28dBc adjacent channel power ratio (ACPR) compared with the traditional peak windowing clipping methods in situation that peak is maximally suppressed. This technique also has low algorithm complexity so it can be easily laid out on hardware. The proposed algorithm has been laid out on a low-cost field-programmable gate array (FPGA) to verify its effectiveness and feasibility. A 64-QAM modulated 20M LTE-A waveform is used for measurement, which has a sampling rate of 245.67M.

Sparse orthogonal matrices are proposed to improve the convergence property of expectation propagation (EP) for sparse signal recovery from compressed linear measurements subject to known dense and ill-conditioned multiplicative noise. As a typical problem, this letter addresses generalized spatial modulation (GSM) in over-loaded and spatially correlated multiple-input multiple-output (MIMO) systems. The proposed sparse orthogonal matrices are used in precoding and constructed efficiently via a generalization of the fast Walsh-Hadamard transform. Numerical simulations show that the proposed sparse orthogonal precoding improves the convergence property of EP in over-loaded GSM MIMO systems with known spatially correlated channel matrices.

Part based models have been proved to be beneficial for person re-identification (Re-ID) in recent years. Existing models usually use fixed horizontal stripes or rely on human keypoints to get each part, which is not consistent with the human visual mechanism. In this paper, we propose a Self-Channel Attention Weighted Part model (SCAWP) for Re-ID. In SCAWP, we first learn a feature map from ResNet50 and use 1x1 convolution to reduce the dimension of this feature map, which could aggregate the channel information. Then, we learn the weight map of attention within each channel and multiply it with the feature map to get each part. Finally, each part is used for a special identification task to build the whole model. To verify the performance of SCAWP, we conduct experiment on three benchmark datasets, including CUHK03-NP, Market-1501 and DukeMTMC-ReID. SCAWP achieves rank-1/mAP accuracy of 70.4%/68.3%, 94.6%/86.4% and 87.6%/76.8% on three datasets respectively.

Offset quadrature amplitude modulation based orthogonal frequency division multiplexing (OFDM/OQAM) is a promising multi-carrier modulation technique to achieve a low-sidelobe spectrum while maintaining orthogonality among subcarriers. However, a major shortcoming of OFDM/OQAM systems is the high peak-to-average power ratio (PAPR) of the transmit signal. To resolve the high-PAPR issue of traditional OFDM, a self-synchronized-scrambler-based selected-mapping has been investigated, where the transmit sequence is scrambled to reduce PAPR. In this method, the receiver must use a descrambler to recover the original data. However, the descrambling process leads to error propagation, which degrades the bit error rate (BER). As described herein, a partial scrambling overlapped selected mapping (PS-OSLM) scheme is proposed for PAPR reduction of OFDM/OQAM signals, where candidate sequences are generated using partial scrambling of original data. The best candidate, the one that minimizes the peak amplitude within multiple OFDM/OQAM symbols, is selected. In the proposed method, an overlap search algorithm for SLM is applied to reduce the PAPR of OFDM/OQAM signals. Numerical results demonstrate that our PS-OSLM proposal achieves better BER than full-scrambling overlapped SLM (FS-OSLM) in OFDM/OQAM systems while maintaining almost equivalent PAPR reduction capability as FS-OSLM and better PAPR than SLM without overlap search. Additionally, we derive a theoretical lower bound expression for OFDM/OQAM with PS-OSLM, and clarify the effectiveness of the proposed scheme.

To reduce peak-to-average power ratio, we propose a method of choosing suitable vectors in a partial transmit sequence technique. Conventional approaches require that a suitable vector be selected from a large number of candidates. By contrast, our method does not include such a selecting procedure, and instead generates random vectors from the Gaussian distribution whose covariance matrix is a solution of a relaxed problem. The suitable vector is chosen from the random vectors. This yields lower peak-to-average power ratio than a conventional method.

The application of network function virtualization (NFV) and software-defined networking (SDN) to passive optical networks (PONs) is attracting attention for the deployment of cost-effective access network systems. This paper presents a novel architecture of a programmable finite state machine (P-FSM) as a hardware accelerator for protocol processing in an optical line terminal (OLT). The P-FSM is programmable hardware that manages various types of FSMs to enhance flexibility in OLTs and achieve wired-rate performance with a negligible increase in total chip area. The P-FSM is implemented using three key technologies: a specific architecture for state management of communications protocols to minimize the logic area, a memory distributed implementation to minimize the program memory, and a new branch operation to minimize the memory area and reduce processing time. Evaluation results show that the P-FSM can handle 10G-EPON/NG-PON2 communications protocols in the same architecture while achieving wired-rate performance. The increase in the total designed area is only 1.5% to 4.9% depending on the number of protocols supported compared to the area of a conventional communications SoC without flexibility. We also clarify that our architecture has the scalability needed to modify the number of FSMs and the maximum number of ONU connections according to the system scale.

This paper presents a 120-GHz-band split ring resonator (SRR) bandstop filter whose insertion loss can be controlled by coupling another lattice pattern substrate. The SRR bandstop filter and lattice pattern substrate is composed of 200-µm-thick quartz substrate and 5-µm-thick gold patterns. S21 of the SRR bandstop filter is -37.8 dB, and its -10-dB bandwidth is 115-130 GHz. S21 of the SRR bandstop filter changes to -4.1 dB at 125 GHz by arranging the lattice pattern substrate in close proximity to the SRR stopband filter, because coupling between the SRR and the lattice pattern occurs when the SRR and lattice pattern are opposed in close proximity. It was found that 10 Gbit/s data transmission can be achieved by setting the lattice pattern substrate just above the SRR bandstop filter with a spacer thickness of 50 µm, even though data transmission is impossible when only the SRR bandstop filter is inserted between the transmitter and the receiver.

The performance of online algorithms for the bin packing problem is usually measured by the asymptotic approximation ratio. However, even if an online algorithm is explicitly described, it is in general difficult to obtain the exact value of the asymptotic approximation ratio. In this paper we show a theorem that gives the exact value of the asymptotic approximation ratio in a closed form when the item sizes and the online algorithm satisfy some conditions. Moreover, we demonstrate that our theorem serves as a powerful tool for the design of online algorithms combined with mathematical optimization.

This paper deals with the problem of enumerating 3-edge-connected spanning subgraphs of an input plane graph. In 2018, Yamanaka et al. proposed two enumeration algorithms for such a problem. Their algorithm generates each 2-edge-connected spanning subgraph of a given plane graph with n vertices in O(n) time, and another one generates each k-edge-connected spanning subgraph of a general graph with m edges in O(mT) time, where T is the running time to check the k-edge connectivity of a graph. This paper focuses on the case of the 3-edge-connectivity in a plane graph. We give an algorithm which generates each 3-edge-connected spanning subgraph of the input plane graph in O(n2) time. This time complexity is the same as the algorithm by Yamanaka et al., but our algorithm is simpler than theirs.

Let G11 (resp., G12) be the ternary Golay code of length 11 (resp., 12). In this letter, we investigate the separating redundancies of G11 and G12. In particular, we determine the values of sl(G11) for l = 1, 3, 4 and sl(G12) for l = 1, 4, 5, where sl(G11) (resp., sl(G12)) is the l-th separating redundancy of G11 (resp., G12). We also provide lower and upper bounds on s2(G11), s2(G12), and s3(G12).

This paper proposes non-orthogonal packet access based on low density signature with phase only adaptive precoding. The proposed access allows multiple user terminals to send their packets simultaneously for implementing massive connectivity, though only one antenna is put on every terminal and on an access point. This paper proposes a criterion that defines the optimum rotation angles for the phase only precoding, and an algorithm based on the steepest descent to approach the optimum rotation angles. Moreover, this paper proposes two complexity-reduced algorithms that converge much faster than the original algorithm. When 6 packets are transmitted in 4 time slots, i.e., overloading ratio of 1.5, the proposed adaptive precoding based on all the proposed algorithms attains a gain of about 4dB at the BER of 10-4 in Rician fading channels.

Locally repairable codes (LRCs) are a type of new erasure codes designed for modern distributed storage systems (DSSs). In order to obtain ternary LRCs of distance 6, firstly, we propose constructions with disjoint repair groups and construct several families of LRCs with 1 ≤ r ≤ 6, where codes with 3 ≤ r ≤ 6 are obtained through a search algorithm. Then, we propose a new method to extend the length of codes without changing the distance. By employing the methods such as expansion and deletion, we obtain more LRCs from a known LRC. The resulting LRCs are optimal or near optimal in terms of the Cadambe-Mazumdar (C-M) bound.

In this paper, we propose a real-time BSS (Blind Source Separation) system with two microphones that extracts only desired sound sources. Under the assumption that the desired sound sources are close to the microphones, the proposed BSS system suppresses distant sound sources as undesired sound sources. We previously developed a BSS system that can estimate the distance from a microphone to a sound source and suppress distant sound sources, but it was not a real-time processing system. The proposed BSS system is a real-time version of our previous BSS system. To develop the proposed BSS system, we simplify some BSS procedures of the previous system. Simulation results showed that the proposed system can effectively suppress the distant source signals in real-time and has almost the same capability as the previous system.

Given a box of some specified size and a number of pieces of some specified shape, the anti-slide problem considers how to pack the pieces such that none of the pieces in the box can slide in any direction. The object is to find such a sparsest packing. In this paper, we consider the problem for the case of a two-dimensional square box using T-tetromino pieces. We show that, for a square box of side length n, the number of pieces in a sparsest packing is exactly $lfloor 2n/3 floor$ when n≢0 (mod 3), and is between 2n/3-1 and n-1 when n≡0 (mod 3).

This paper describes a method to control prosodic features using phonetic and prosodic symbols as input of attention-based sequence-to-sequence (seq2seq) acoustic modeling (AM) for neural text-to-speech (TTS). The method involves inserting a sequence of prosodic symbols between phonetic symbols that are then used to reproduce prosodic acoustic features, i.e. accents, pauses, accent breaks, and sentence endings, in several seq2seq AM methods. The proposed phonetic and prosodic labels have simple descriptions and a low production cost. By contrast, the labels of conventional statistical parametric speech synthesis methods are complicated, and the cost of time alignments such as aligning the boundaries of phonemes is high. The proposed method does not need the boundary positions of phonemes. We propose an automatic conversion method for conventional labels and show how to automatically reproduce pitch accents and phonemes. The results of objective and subjective evaluations show the effectiveness of our method.