The feasibility condition of interference alignment (IA) for multiple-input multiple-output two-way interference channel is studied in this paper. A necessary condition and a sufficient condition on the IA feasibility are established and the sum degrees of freedom (DoF) for a broad class of network topologies is characterized. The numerical results demonstrate that two-way operation with appropriate IA is able to achieve larger sum DoF than the conventional one-way operation.

Compared to the traditional functional dependency (FD), the extended conditional functional dependency (CFD) has shown greater potential for detecting and repairing inconsistent data. CFDMiner is a widely used algorithm for mining constant-CFDs. But the search space of CFDMiner is too large, and there is still room for efficiency improvement. In this paper, an efficient pruning strategy is proposed to optimize the algorithm by reducing the search space. Both theoretical analysis and experiments have proved the optimized algorithm can produce the consistent results as the original CFDMiner.

Pseudorandom binary sequences balanced and with optimal autocorrelation have many applications in the stream cipher, communication, coding theory, etc. Constructing a binary sequences with three-level autocorrelation is equivalent to finding the corresponding characteristic set of the sequences that should be an almost difference set. In the work of T.W. Cusick, C. Ding, and A. Renvall in 1998, the authors gave the necessary and sufficient conditions by which a set of octic residues modulo an odd prime forms an almost difference set. In this paper we show that no integers verify those conditions by the theory of generalized Pell equations. In addition, by relaxing the definition of almost difference set given by the same authors, we could construct two classes of modified almost difference sets and two ones of difference sets from the set of octic residues.

For the realization of a high-efficiency antenna for 60GHz-band wireless personal area network, we propose placing a CMOS RF circuit and an antenna on opposing sides of a silicon chip. They are connected with low loss by a coaxial-line structure using a hole opening in the chip. Since the CMOS circuit is driven differentially, a differential-feed antenna is used. In this paper, we design and measure a differential-feed square patch antenna on a silicon chip. To enhance the radiation efficiency, it is placed on a 200µm thick resin layer. The calculated radiation efficiency of 79% includes the connection loss. A prototype antenna is measured in a reverberation chamber, and its radiation efficiency is estimated to be about 81±3%.

We herein describe an autonomous peer discovery scheme for Device-to-Device (D2D) communications. With the increasing popularity of D2D communications, an efficient means of finding the neighboring node, i.e., peer discovery, is required. To this end, we propose a new autonomous peer discovery scheme that uses azimuth spread (AS), delay spread (DS), and shadow fading of the uplink pilot from each mobile station (MS). Given that AS, DS, and shadow fading are spatially correlated, nodes that have similar values must be neighbors. The proposed scheme filters out the MSs that are unlikely to be neighbors and uses the Kolmogorov-Smirnov (K-S) test to improve the accuracy of neighbor discovery. Unlike previous peer discovery schemes that incur additional signaling overheads, our proposal finds neighboring nodes by using the existing uplink pilot transmission from MSs such that neighboring peers can be found autonomously. Through analysis and simulation, we show that neighboring MSs can be found accurately with low latency.

An analog-beamforming-based eigenmode transmission technique is proposed that employs a network of interconnected 180-degree hybrid couplers at both transmitting and receiving sides of a plane-symmetrically configured short-range MIMO system. This technique can orthogonalize MIMO channels regardless of array parameters such as antenna spacing and Tx-Rx distance, provided the MIMO array is symmetric. For verifying the effectiveness of the proposed technique in channel orthogonalization, an experiment is conducted using a 4×4 MIMO array consisting of microstrip antennas and cascade-connected rat-race hybrid couplers. The results indicate a reduction in interference by approximately -28.3dB on average compared to desired signal power, and the ability to realize four-stream parallel MIMO transmission by using only analog passive networks. The proposed technique can achieve channel capacity almost equivalent to that of eigenbeam space division multiplexing with ideal digital beamforming.

This paper proposes Fast Fourier Transform (FFT) based orthogonal beam selection method at the user terminals (UTs) to reduce the number of nulls for the other users except an intended user by the Block Diagonalization (BD) algorithm in multiuser MIMO (MU-MIMO) sytems. The BD algorithm has been proposed in order to realize MU-MIMO broadcast transmission with a realistic signal processing burden. The BD algorithm cancels inter-user interference by creating the weights so that the channel matrixes for the other users are set to be zero matrixes. However, when the number of transmit antennas is equals to the total number of received antennas, the transmission rate by the BD algorithm is decreased. The proposed method realizes the performance improvement compared to the conventional BD algorithm without the burden on the UTs. It is verified via bit error rate (BER) evaluation that the proposed method is effective compared to the conventional BD algorithm and antenna selection method. Moreover, the effectiveness of proposed method is verified by the performance evaluation considering medium access control (MAC) layer in a comparison with the conventional BD algorithm which needs the channel state information (CSI) feedback. Because the proposed method can be easily applied to beamforming without the CSI feedback (implicit beamforming), it is shown that the propose method is effective from a point of view on the transmission efficiency in MU-MIMO system.

A regulated charge pump (CP) with an extended range of load current is presented. A power-efficient adaptive feedback controller is adopted. Verified by a 0.18µm CMOS technology with a power supply of 3.3V, the measured output voltage of the CP is regulated above 5V when the load current is varied from 2.5mA to 50mA. The measured power efficiency spans from 81.7% at lighter load to 75.2% when load current is 50mA. The measured output ripples are small and below 24mV.

Information Centric Networking (ICN) is a promising architecture as an alternative paradigm to traditional IP networking. The innovative concepts, such as named data, name-based routing, and in-network caching bring lots of benefits to Wireless Sensor Networks (WSNs). Simple and robust communication model of ICN, based on interest/data messages exchange, is appealing to be deployed in WSNs. However, ICN architectures are designed for power supplied network devices rather than resource-constrained sensor nodes. Introducing ICN-liked architecture to WSNs needs to rethink the naming scheme and forwarding strategy to meet the requirements of energy efficiency and failure recovery. This paper presents a light weight data centric routing mechanism (GRMR) for interest dissemination and data delivery in location-aware WSNs. A simple naming scheme gives assistance for routing decision by individual nodes. Greedy routing engaging with regional multicast mechanism provides an efficient data centric routing approach. The performance is analytically evaluated and simulated in NS-2. The results indicate that GRMR achieves significant energy efficiency under investigated scenarios.

EPCglobal architecture framework is divided into identify, capture, and share layers and defines a collection of standards. It is not fully adequate to build IoT applications because the transducer capability is lacking. IEEE 1451 is a set of standards that defines data exchange format, communication protocols, and various connection interfaces between sensors/actuators and transducer interface modules. By appending IEEE 1451 transducer capability to EPCglobal architecture framework, a consistent EPC scheme expression for heterogeneous things can be achieved at identify layer. It is benefit to extend the upper layers of EPCglobal architecture framework seamlessly. In this paper, we put our emphasis on how to leverage the transducer capability at the capture layer. A device cycle, transducer cycle specification, and transducer cycle report are introduced to collect and process sensor/actuator data. The design and implementation of GS1 EPCglobal Application Level Events (ALE) modules extension are proposed for explaining the design philosophy and verifying the feasibility. It will interact with the capture and query services of EPC Information Services (EPCIS) for IoT applications at the share layer. By cooperating and interacting with these layers of EPCglobal architecture framework, the IoT architecture EPCglobal+ based on international standards is built.

This paper explores the potential of pitch determination from bone conducted (BC) speech. Pitch determination from normal air conducted (AC) speech signal can not attain the expected level of accuracy for every voice and background conditions. In contrast, since BC speech is caused by the vibrations that have traveled through the vocal tract wall, it is robust against ambient conditions. Though an appropriate model of BC speech is not known, it has regular harmonic structure in the lower spectral region. Due to this lowpass nature, pitch determination from BC speech is not usually affected by the dominant first formant. Experiments conducted on simultaneously recorded AC and BC speech show that BC speech is more reliable for pitch estimation than AC speech. With little human work, pitch contour estimated from BC speech can also be used as pitch reference that can serve as an alternate to the pitch contour extracted from laryngograph output which is sometimes inconsistent with simultaneously recorded AC speech.

Reversible data hiding is a technique in which hidden data are embedded in host data such that the consistency of the host is perfectly preserved and its data are restored during extraction of the hidden data. In this paper, a linear prediction technique for reversible data hiding of audio waveforms is improved. The proposed variable expansion method is able to control the payload size through varying the expansion factor. The proposed technique is combined with the prediction error expansion method. Reversible embedding, perfect payload detection, and perfect recovery of the host signal are achieved for a framed audio signal. A smaller expansion factor results in a smaller payload size and less degradation in the stego audio quality. Computer simulations reveal that embedding a random-bit payload of less than 0.4 bits per sample into CD-format music signals provide stego audio with acceptable objective quality. The method is also applied to G.711 µ-law-coded speech signals. Computer simulations reveal that embedding a random-bit payload of less than 0.1 bits per sample into speech signals provide stego speech with good objective quality.

A novel directional coupler loaded with feedback capacitances on the coupled lines is presented. Its effect of enhancing the coupling is qualitatively shown by deriving an equation for the coupling. Besides, a method to compensate for the phase difference between the even and odd modes of the coupler is presented. To demonstrate, a novel tandem 3-dB coupler consisting of the proposed coupled lines is designed and described. In addition, a waveguide (rectangular coaxial line) 8×8 HYB matrix using planar double-layer structure that is composed of the proposed tandem 3-dB couplers and branch-line couplers, which is operated in S-band, is designed and fabricated showing excellent performance.

Two spanning trees T1,T2 of a graph G = (V,E) are independent if they are rooted at the same vertex, say r, and for each vertex v ∈ V, the path from r to v in T1 and the path from r to v in T2 have no common vertices and no common edges except for r and v. In general, spanning trees T1,T2,…,Tk of a graph G = (V,E) are independent if they are pairwise independent. A graph G = (V,E) is called a 2-chordal ring and denoted by CR(N,d1,d2), if V = {0,1,…,N-1} and E = {(u,v)|[v-u]N = 1 or [v-u]N = d1 or [v-u]N = d2, 2 ≤ d1 < d2 ≤ N/2}. CR(N,d1,N/2) is 5-connected if N ≥ 8 is even and d1 ≠ N/2-1. We give an algorithm to construct 5 independent spanning trees of CR(N,d1,N/2),N ≥ 8 is even and 2 ≤ d1 ≤ ⌈N/4⌉.

A 12×16-element corporate-feed slot array is presented. The corporate-feed circuit for the 12×16-elemtent array consists of cross-junctions and asymmetric T-junctions, whereas the conventional one is limited to arrays of 2m×2n slots by its use of symmetric T-junctions. Simulations of the 12×16-element array show a 7.6% bandwidth for reflection less than -14dB. A 31.7-dBi gain with an antenna efficiency of 82.6% is obtained at the design frequency of 61.5GHz. The 12×16-element array is fabricated by diffusion bonding of laminated thin metal plates. Measurements indicate 31.1-dBi gain with 71.9% antenna efficiency at 61.5GHz.

In this letter, an effective acoustic feedback cancellation algorithm is proposed based on the normalized sub-band adaptive filter (NSAF). To improve the confliction between fast convergence rate and low misalignment in the NSAF algorithm, a variable step size is designed to automatically vary according to the update state of the filter. The update state of the filter is adaptively detected via the normalized distance between the long term average and the short term average of the tap-weight vector. Simulation results demonstrate that the proposed algorithm has superior performance in terms of convergence rate and misalignment.

This paper presents a new methodology of the over-the-air (OTA) assessment for vertically arranged multiple-input multiple-output (MIMO) array antennas. Particular emphasis is placed on how well handset MIMO antennas with a vertically arranged structure are characterized using the limited number of scatterers implemented in a fading emulator. First we studied the mechanism of the arrangement of scatterers on the variation of channel responses using a proposed three-dimensional analytical model. It is shown that the condition of a 3D-OTA with the prescribed parameters allows the correlation to be reduced, which permits the channel capacity to increase in the same manner that sufficient scatterers are distributed over the entire solid angle. Then the appropriate scatterers arrangement for a 3D-OTA instrument considering the number of DUT antenna elements and multipath characteristics is investigated. The analytical results show that a suitable scatterers arrangement can be determined for various conditions of multipath environments and numbers of array elements, and that the arrangement can be employed for designing an actual 3D-OTA apparatus.

This letter describes a speech enhancement algorithm for stereo signals corrupted by diffuse noise. It estimates the noise signal and also a beamformed target signal based on blind target signal cancelation derived from sparsity minimization. Enhanced target speech is obtained by Wiener filtering using both the signals. Experimental results demonstrate the effectiveness of the proposed method.

This paper presents the first non-trivial collision attack on the double-block-length compression function presented at FSE 2006 instantiated with round-reduced AES-256: f0(h0||h1,M)||f1(h0||h1,M) such that f0(h0||h1, M) = Eh1||M(h0)⊕h0 , f1(h0||h1,M) = Eh1||M(h0⊕c)⊕h0⊕c , where || represents concatenation, E is AES-256 and c is a 16-byte non-zero constant. The proposed attack is a free-start collision attack using the rebound attack proposed by Mendel et al. The success of the proposed attack largely depends on the configuration of the constant c: the number of its non-zero bytes and their positions. For the instantiation with AES-256 reduced from 14 rounds to 8 rounds, it is effective if the constant c has at most four non-zero bytes at some specific positions, and the time complexity is 264 or 296. For the instantiation with AES-256 reduced to 9 rounds, it is effective if the constant c has four non-zero bytes at some specific positions, and the time complexity is 2120. The space complexity is negligible in both cases.

Recognition of second language (L2) speech is still a challenging task even for state-of-the-art automatic speech recognition (ASR) systems, partly because pronunciation by L2 speakers is usually significantly influenced by the mother tongue of the speakers. The authors previously proposed using a reduced phoneme set (RPS) instead of the canonical one of L2 when the mother tongue of speakers is known, and demonstrated that this reduced phoneme set improved the recognition performance through experiments using English utterances spoken by Japanese. However, the proficiency of L2 speakers varies widely, as does the influence of the mother tongue on their pronunciation. As a result, the effect of the reduced phoneme set is different depending on the speakers' proficiency in L2. In this paper, the authors examine the relation between proficiency of speakers and a reduced phoneme set customized for them. The experimental results are then used as the basis of a novel speech recognition method using a lexicon in which the pronunciation of each lexical item is represented by multiple reduced phoneme sets, and the implementation of a language model most suitable for that lexicon is described. Experimental results demonstrate the high validity of the proposed method.