The dramatic growth in wireless data traffic has triggered the investigation of fifth generation (5G) wireless communication systems. Small cells will play a very important role in 5G to meet the 5G requirements in spectral efficiency, energy savings, etc. In this paper, we investigate low complexity millimeter-wave communication systems with uniform circular arrays (UCAs) in line-of-sight (LOS) multiple-input multiple-output (MIMO) channels, which are used in fixed wireless access such as small cell wireless backhaul for 5G. First, we demonstrate that the MIMO channel matrices for UCAs in LOS-MIMO channels are circulant matrices. Next, we provide a detailed derivation of the unified optimal antenna placement which makes MIMO channel matrices orthogonal for 3×3 and 4×4 UCAs in LOS channels. We also derive simple analytical expressions of eigenvalues and capacity as a function of array design (link range and array diameters) for the concerned systems. Finally, based on the properties of circulant matrices, we propose a high performance low complexity LOS-MIMO precoding system that combines forward error correction (FEC) codes and spatial interleaver with the fixed IDFT precoding matrix. The proposed precoding system for UCAs does not require the channel knowledge for estimating the precoding matrix at the transmitter under the LOS condition, since the channel matrices are circulant ones for UCAs. Simulation results show that the proposed low complexity system is robust to various link ranges and can attain excellent performance in strong LOS environments and channel estimation errors.

In the near future, interconnection networks of massively parallel computer systems will connect more than a hundred thousands of computing nodes. The performance evaluation of the interconnection networks can provide real insights to help the development of efficient communication library. Hence, to evaluate the performance of such interconnection networks, simulation tools capable of modeling the networks with sufficient details, supporting a user-friendly interface to describe communication patterns, providing the users with enough performance information, completing simulations within a reasonable time, are a real necessity. This paper introduces a novel interconnection network simulator NSIM, for the evaluation of the performance of extreme-scale interconnection networks. The simulator implements a simplified simulation model so as to run faster without any loss of accuracy. Unlike the existing simulators, NSIM is built on the execution-driven simulation approach. The simulator also provides a MPI-compatible programming interface. Thus, the simulator can emulate parallel program execution and correctly simulate point-to-point and collective communications that are dynamically changed by network congestion. The experimental results in this paper showed sufficient accuracy of this simulator by comparing the simulator and the real machine. We also confirmed that the simulator is capable of evaluating ultra large-scale interconnection networks, consumes smaller memory area, and runs faster than the existing simulator. This paper also introduces a simulation service built on a cloud environment. Without installing NSIM, users can simulate interconnection networks with various configurations by using a web browser.

The paper argues that a radical shift in the market for communications services is emerging, driven by the mass availability of cheap bandwidth, computing and global mobility combined with the pervasive rise of Internet based data services. At the same time, the Operation Support Systems (OSS's*) that are essential in order to create business value from these technologies are lagging behind market need. The authors argue for a re-think of the humble management system into a complete software wrap-around of the network to deliver a value creation platform - as different from yesterday's OSS as the bakelite telephone is from today's tri-band mobile handsets. This software will be based on product standards, not paper ones and will require a major shift of gears from the position of today. This value creation platform will be built from advanced, component based software delivered through a very different market model to that of today. Much of this technology exists; we simply need critical mass behind a common approach. The discussion in this paper represents the personal views of the authors and does not necessarily represent the views of any organisation.

Telecommunications management activities have mostly been supported by operators; however, machines are gradually playing more important roles in the management arena by utilizing computing technology. Additionally, management systems can now be networked by using standard interface specifications. The study of human and machine integration is thus essential for achieving the sophisticated management objectives of telecommunications. This paper proposes the principles for a telecommunications management integration network (TMIN), which integrates human and machine management networks, and proposes a source text description method for transferring management communication knowledge from human to machine. First, reference models are proposed for the management process and management communication. These models cover network management activities of both humans and machines. Second, the contents of the source text are clarified. Source text presents human management knowledge in a form suitable for machine-machine communication. Third, an efficient source text description method is proposed that reduces redundancy and proliferation. Finally, a means of harmonizing management information definitions with TMIN is suggested to facilitate human-machine cooperation.

Telecommunications management is essentially an aggregation of a wide range of activities, including operations and management (O & M) of telecommunications services and customers as well as the network and network elements. During the period of rapid growth in telecommunications, the highest priority was to meet increasing market demand and to construct a telecommunications network infrastructure. Therefore, research and development in telecommunications management were subordinated to the evolution of telecommunications services and systems. Recently, customers have been demanding higher quality services, as a variety of new, advanced services have been introduced. This has led to the need to integrate telecommunications services and O & M services. This paper first reviews the history of the development of telecommunications O & M in parallel with the progress of telecommunications in Japan and clarifies specific features in each step of this progress. This analysis identifies urgent problems and their solutions. The results suggests that telecommunications O & M and O & M services should be considered as a key to making future services possible and to the value of those services in a multi-media telecommunications services environment. Based on these studies, the future direction of O & M is then shown, focusing on cooperative O & M involving the customer in the multi-media, multidomain telecommunications environment.

We present an attractive approach for OFDM transmission using an adaptive pre-FFT equalizer, which can select ICI reduction mode according to channel condition, and a degenerated-inverse-matrix-based channel estimator (DIME), which uses a cyclic sinc-function matrix uniquely determined by transmitted subcarriers. In addition to simulation results, the proposed system with an adaptive pre-FFT equalizer and DIME has been laboratory tested by using a software defined radio (SDR)-based test bed. The simulation and experimental results demonstrated that the system at a rate of more than 100 Mbps can provide a bit error rate of less than 10-3 for a fast multi-path fading channel that has a moving velocity of more than 200 km/h with a delay spread of 1.9 µs (a maximum delay path of 7.3 µs) in the 5-GHz band.

This paper presents a periodic sampling queueing model with (active) processing/interruption phases, Poisson arrivals and a generalized processing probability in accordance with the number of customers at sampling points. The generating functions of the number of customers in a system immediately following sampling points are obtained from the existing results of a generalized periodic sampling model. Using these functions, generating functions of the number of customers at arbitrary points are formulated for active processing and interruption phases. Performance measures such as average customers in a system and average waiting time are derived. Formulas for a single processor model are tailored for a specific application to a common channel signaling system shared between call signals and database information signals. Numerical examples for this system are presented as a guide for the traffic design regarding the ratios of active/interruption phaese and sampling preiod/average processing time.

This paper presents an analog 2-dimensional discrete cosine transform (2-D DCT) processor for focal-plane image compression. The on-chip analog 2-D DCT processor can process directly the analog signal of the CMOS image sensor. The analog-to-digital conversion (ADC) is preformed after the 2-D DCT, and this leads to efficient AD conversion of video signals. Most of the 2-D DCT coefficients can be digitized by a relatively low-resolution ADC or a zero detector. The quantization process after the 2-D DCT can be realized by the ADC at the same time. The 88-point analog 2-D DCT processor is designed by switched-capacitor (SC) coefficient multipliers and an SC analog memory based on 0.35µm CMOS technology. The 2-D DCT processor has sufficient precision, high processing speed, low power dissipation, and small silicon area. The resulting smart image sensor chips with data compression and digital transmission functions are useful for the high-speed image acquisition devices and portable digital video camera systems.

A new BiCMOS process based on a high-speed bipolar process with 0.5 µm emitter width has been developed using a bonded SOI substrate. Double polysilicon bipolar transistors with the trench isolation, shallow junctions and the pedestal collector implantation provide a high cut-off frequency of 27 GHz. Stress induced device degradation is carefully examined and a low stress trench isolation process is proposed.

This paper provides a performance evaluation of our audio and video CODEC by using a method for rapidly verifying and evaluating overall performance on real-time workloads of system LSIs integrated with SPXK5SC DSP cores. The SPXK5SC have been developed as a DSP core well-suited to system LSIs. Despite the fact that it is very important to evaluate the overall performance of target LSIs on real workloads before actual LSI fabrication, software simulators are too slow to deal with real workloads and full hardware prototyping is unable to respond well to design improvements. Therefore, we have developed a hardware emulation approach to be used on system LSIs integrated with a SPXK5SC DSP core in order to evaluate the overall performance of audio/video CODEC on a target system. Our emulation system using a DSP core TEG, which has a bus interface, and an FPGA is suitable for overall system evaluation on real-time workloads as well as architectural investigation. In this paper, we discuss the use of the emulation system in evaluating performance during AV CODEC execution. In addition, an architecture design based on our emulation system is also described.

This paper presents a general model and analysis of periodically sampled queues, which cover a variety of real-time processing systems such as TSS computers and telecommunication switching systems. Generating functions of the number of calls in a system immediately following sampling points are formulated under the conditions of; (1) general sampling period distribution, (2) independent, identically distributed arrivals per sampling interval, (3) maximum S processing capability per sampling interval, and (4) service probability per sampling interval in accordance with the number of calls in the system immediately following sampling points. Thus, probability distributions of the number of calls at arbitrary instants are formulated as generating functions. With these, performance measures such as the average number of calls in a system and the average waiting time are derived. The universality of this model is demonstrated by special cases. Numerical examples are given to examine the effects of sampling distributions under a single server condition, taken as an example.

Wireless sensor networks play an important role in several industries. Ad-hoc networks with multi-hop transmissions are considered suitable for wireless sensor networks because of their high scalability and low construction cost. However, a lack of centralized control makes it difficult to respond to congestion when system capacity is exceeded. Therefore, the analysis of system capacity is a critical issue for system design. In this paper, we propose a novel zone division model to analyze the capacity of multi-hop wireless sensor networks using carrier sense multiple access with collision avoidance protocols. We divide the one-hop area to a gateway (GW) into two zones: an outer zone, where access nodes (ANs) can relay packets from multi-hop ANs, and an inner zone where ANs cannot relay packets. Using this approach, we calculate the packet loss for each zone to estimate the capacity, considering the difference in the communication range of the GW and ANs, as well as the collision with hidden nodes. Comparisons with simulation results and the conventional method show that our model achieves higher estimation accuracy.