Jaehak CHUNG Seung Hoon NAM Chan-Soo HWANG
High Rate Space-Time Block Codes (HR-STBCs) with greater than 1 symbol/transmission and simple decoding schemes are proposed. The HR-STBC demonstrates 3 dB Eb/No gain at BER = 10-3 compared with the conventional STBC when three transmit antennas and two receive antennas are utilized.
Seung Hoon NAM Jaehak CHUNG Chan-Soo HWANG
A design of non-orthogonal 33 space-time block code (STBC) is proposed. The proposed design achieves full rate, full level diversity, and maximum coding gain by symbol rotation (SR) method. In addition, the proposed scheme has lower encoding complexity than the unitary constellation-rotation (CR) STBC, while two methods exhibit the same bit error rate (BER) performance in Rayleigh fading channel.
Yu LIU Satoshi KOMATSU Masahiro FUJITA
Recently, system level design languages (SLDL), which can describe both hardware and software aspects of the design, are receiving attention. Mixed-signal extensions of SLDL enable current discrete-oriented SLDLs to describe and simulate not only digital systems but also digital-analog mixed-signal systems. The synchronization between discrete and continuous behaviors is widely regarded as a critical part in the extensions. In this paper, we present an event-driven synchronization mechanism for both timed and untimed system level designs through which discrete and continuous behaviors are synchronized via AD events and DA events. We also demonstrate how the synchronization mechanism can be incorporated into the kernel of SLDL, such as SpecC. In the extended kernel, a new simulation cycle, the AMS cycle, is introduced. Three case studies show that the extended SpecC-based system level design environment using our synchronization mechanism works well with timed/untimed mixed-signal system level description.
Masakazu ADACHI Toshimitsu USHIO
This paper analyzes automation surprises in human-machine systems with time information. Automation surprises are phenomena such that the underlying machine's behavior diverges from user's intention and may lead to critical situations. Thus, designing human-machine systems without automation surprises is one of fundamental issues to achieve reliable user interaction with the machines. In this paper, we focus on timed human-machine interaction and address their formal aspects. The presented framework is essentially an extension of untimed human-machine interaction and will cover the previously proposed methodologies. We employ timed automata as a model of human-machine systems with time information. Modeling the human-machine systems as timed automata enables one to deal with not only discrete behavior but also time constraints. Then, by introducing the concept of timed simulation of the machine model and the user model, conditions which guarantee the nonexistence of automation surprises are derived. Finally, we construct a composite model in which a machine model and a user model evolve concurrently and show that automation surprises can be detected by solving a reachability problem in the composite model.
Since their inception almost fifty years ago, hidden Markov models (HMMs) have have become the predominant methodology for automatic speech recognition (ASR) systems--today, most state-of-the-art speech systems are HMM-based. There have been a number of ways to explain HMMs and to list their capabilities, each of these ways having both advantages and disadvantages. In an effort to better understand what HMMs can do, this tutorial article analyzes HMMs by exploring a definition of HMMs in terms of random variables and conditional independence assumptions. We prefer this definition as it allows us to reason more throughly about the capabilities of HMMs. In particular, it is possible to deduce that there are, in theory at least, no limitations to the class of probability distributions representable by HMMs. This paper concludes that, in search of a model to supersede the HMM (say for ASR), rather than trying to correct for HMM limitations in the general case, new models should be found based on their potential for better parsimony, computational requirements, and noise insensitivity.
Dianjun CHEN Takeshi HASHIMOTO
We propose two sequence design schemes for an overloaded space-time spreading system with multiple antennas. One scheme is for a system in which the amplitude of user signals needs not be adjusted and provides tradeoffs between the user capacity and diversity order. This scheme has a certain similarity to time-sharing, but its performance is further improved by time-diversity. Another is to achieve full diversity order by varying user signal amplitudes. The diversity orders of the respective schemes are theoretically proved and their performances are demonstrated by simulation.
Yukihito OOWAKI Shinichiro SHIRATAKE Toshihide FUJIYOSHI Mototsugu HAMADA Fumitoshi HATORI Masami MURAKATA Masafumi TAKAHASHI
The module-wise dynamic voltage and frequency scaling (MDVFS) scheme is applied to a single-chip H.264/MPEG-4 audio/visual codec LSI. The power consumption of the target module with controlled supply voltage and frequency is reduced by 40% in comparison with the operation without voltage or frequency scaling. The consumed power of the chip is 63 mW in decoding QVGA H.264 video at 15 fps and MPEG-4 AAC LC audio simultaneously. This LSI keep operating continuously even during the voltage transition of the target module by introducing the newly developed dynamic de-skewing system (DDS) which watches and control the clock edge of the target module.
Hongmei WANG Xiang CHEN Shidong ZHOU Ming ZHAO Yan YAO
In this letter, we propose a partial minimum mean-squared error (MMSE) with successive interference cancellation (PMMSESIC) method in frequency domain to mitigate ICI caused by channel variation. Each detection, the proposed method detects the symbol with the largest received signal-to-interference-plus-noise ratio (SINR) among all the undetected symbols, using an MMSE detector that considers only the interference of several neithborhood subcarriers. Analysis and simulations show that it outperforms the MMSE method at relatively high Eb/N0 and its performance is close to the MMSE with successive detection (MMSESD) method in relatively low Doppler frequency region.
Takashi MORIMOTO Hidekazu ADACHI Osamu KIRIYAMA Tetsushi KOIDE Hans Jurgen MATTAUSCH
This letter presents a boundary-active-only (BAO) power reduction technique for cell-network-based region-growing video segmentation. The key approach is an adaptive situation-dependent power switching of each network cell, namely only cells at the boundary of currently grown regions are activated, and all the other cells are kept in low-power stand-by mode. The effectiveness of the proposed technique is experimentally confirmed with CMOS test-chips having small-scale cell networks of up to 4133 cells, where an average of only 1.7% of the cells remains active after application of the proposed approach. About 85% power reduction is thus achievable without sacrificing real-time processing.
Kang-Yoon LEE Hyunchul KU Young Beom KIM
This paper presents a fast switching CMOS frequency synthesizer with a new coarse tuning method for PHS applications. To achieve the fast lock-time and the low phase noise performance, an efficient bandwidth control scheme is proposed. To change the bandwidth, the charge pump current and the loop filter zero resistor should be changed. Charge pump up/down current mismatches are compensated with the current mismatch compensation block. The proposed coarse tuning method selects the optimal tuning capacitances of the LC-VCO to optimize the phase noise and the lock-time. The measured lock-time is about 20 µs and the phase noise is -121 dBc/ at 600 kHz offset. This chip is fabricated with 0.25 µm CMOS technology, and the die area is 0.7 mm2.1mm. The power consumption is 54 mW at 2.7 V supply voltage.
Toshiaki KOIKE Yukinaga SEKI Hidekazu MURATA Susumu YOSHIDA Kiyomichi ARAKI
We developed two types of practical maximum-likelihood detectors (MLD) for multiple-input multiple-output (MIMO) systems, using a field programmable gate array (FPGA) device. For implementations, we introduced two simplified metrics called a Manhattan metric and a correlation metric. Using the Manhattan metric, the detector needs no multiplication operations, at the cost of a slight performance degradation within 1 dB. Using the correlation metric, the MIMO-MLD can significantly reduce the complexity in both multiplications and additions without any performance degradation. This paper demonstrates the bit-error-rate performance of these MLD prototypes at a 1 Gbps-order real-time processing speed, through the use of an all-digital baseband 44 MIMO testbed integrated on the same FPGA chip.
To realize a secure networking infrastructure, the author is carrying out CUE (Coordinating Users' requirements and Engineering constraints) project with a network carrier and a VLSI manufacture. Since CUE-series data-driven processors developed in the project were specifically designed to be an embedded programmable component as well as a multi-processor element, particular design considerations were taken to achieve real-time multiprocessing capabilities essentially needed in multi-media communication environment. A novel data-driven paradigm is first introduced with special emphasis on VLSI-oriented parallel processing architectures. Data-driven protocol handlings on CUE-p and CUE-v1 are then discussed for their real-time multiprocessing capability without any runtime overheads. The emulation facility RESCUE (Real-time Execution System for CUE-series data-driven processors) was also built to develop scalable chip multi-processors in self-evolutional manner. Based on emulation results, the latest version named CUE-v2 was realized as a hybrid processor enabling simultaneous processing of data-driven and control-driven threads to achieve higher performance for inline processing and to avoid any bottlenecks in sequential parts of real-time programs frequently encountered in actual time-sensitive applications. Effectiveness of the data-driven chip multi-processor architecture will finally be addressed for lower power consumption and scalability to realize future VLSI processors in the sub-100 nm era.
Zhengwei GONG Taiyi ZHANG Haiyuan LIU Feng LIU
Space-time coding (STC) schemes for communication systems employing multiple transmit and receive antennas have received considerable interest recently. On space-time coding, some algorithms with perfect channel state information (CSI) have been proposed. In certain fast varying situation, however, it may be difficult to estimate the channel accurately and it is natural to study the blind detection algorithm without CSI. In this paper, based on subspace, a new blind detection algorithm without CSI is proposed. Using singular value decomposition (SVD) on output signal, noise subspace and signal subspace, which keep orthogonal to each other, are obtained. By searching the intersection of the signal subspace and the limited symbol vector set, symbol detection is achieved. The simulations illustrate that the proposed algorithm significantly improves system performance by receiving more output signals relative to transmit symbols. Furthermore, the presented algorithm is robust to the fading channel that changes between two successive blocks.
Hyeon Chyeol HWANG Seung Hoon SHIN Seok Ho KIM Kyung Sup KWAK
In this letter, we propose adaptive linear detectors in space-time block coded multiuser systems, by exploiting a particular property of the minimum mean square error multiuser detector. The proposed scheme can provide much faster convergence than the existing adaptive scheme [5] and so lower the system overhead requirements.
In this paper, we derive an analytical result for channel holding time distribution in mobile satellite networks under general call holding time distribution.
Takehiro IHARA Takayuki NAGAI Kazuhiko OZEKI Akira KUREMATSU
We present a novel approach for single-channel noise reduction of speech signals contaminated by additive noise. In this approach, the system requires speech samples to be uttered in advance by the same speaker as that of the input signal. Speech samples used in this method must have enough phonetic variety to reconstruct the input signal. In the proposed method, which we refer to as referential reconstruction, we have used a small database created from examples of speech, which will be called reference signals. Referential reconstruction uses an example-based approach, in which the objective is to find the candidate speech frame which is the most similar to the clean input frame without noise, although the input frame is contaminated with noise. When candidate frames are found, they become final outputs without any special processing. In order to find the candidate frames, a correlation coefficient is used as a similarity measure. Through automatic speech recognition experiments, the proposed method was shown to be effective, particularly for low-SNR speech signals corrupted with white noise or noise in high-frequency bands. Since the direct implementation of this method requires infeasible computational cost for searching through reference signals, a coarse-to-fine strategy is introduced in this paper.
Haris GACANIN Shinsuke TAKAOKA Fumiyuki ADACHI
For alleviating the high peak-to-average power ratio (PAPR) problem of orthogonal frequency division multiplexing (OFDM), the OFDM combined with time division multiplexing (TDM) using frequency-domain equalization (FDE) was proposed. In this paper, the theoretical bit error rate (BER) analysis of the OFDM/TDM in a frequency-selective fading channel is presented. The conditional BER expression is derived, based on a Gaussian approximation of the inter-symbol interference (ISI) arising from channel frequency-selectivity, for the given set of channel gains. Various FDE techniques as in multi-carrier code division multiple access (MC-CDMA), i.e., zero forcing (ZF), maximum ratio combining (MRC) and minimum mean square error (MMSE) criteria are considered. The average BER performance is evaluated by Monte-Carlo numerical computation method using the derived conditional BER expression.
Toshiya MASHIMA Takanori FUKUOKA Satoshi TAOKA Toshimasa WATANABE
The 2-vertex-connectivity augmentation problem for a specified set of vertices of a graph with degree constraints, 2VCA-SV-DC, is defined as follows: "Given an undirected graph G = (V,E), a specified set of vertices S ⊆V with |S|3 and a function g:V→Z+∪{∞}, find a smallest set E' of edges such that (V,E ∪E') has at least two internally-disjoint paths between any pair of vertices in S and such that vertex-degree increase of each v ∈V by the addition of E' to G is at most g(v), where Z+ is the set of nonnegative integers." This paper shows a linear time algorithm for 2VCA-SV-DC.
Wireless sensor networks present a promising opportunity for realizing many practical applications. Tracking is one of the important applications of these networks. Many approaches have been proposed in the literature to deal with the tracking problem. Recently, a particular type of tracking problem called on-site tracking has been introduced [15],[16]. On-site tracking has been characterized as the tracking in which the sink is eventually required to be present in the vicinity of the target, possibly to perform further actions. In this paper, first we propose two efficient on-site tracking algorithms. Then, we derive theoretical upper bounds for the tracking time and the number of messages generated by the sensor nodes during the tracking for our algorithms. Finally, we present a simulation study that we conducted to evaluate the performance of our algorithms. The results show that our algorithms are efficient as compared to the other existing methods that can solve the on-site tracking problem. In particular, the path adaptive nature of the sink in our algorithms allows the network to conserve the energy and the sink to reduce the tracking time.
A chordal graph is a graph which contains no chordless cycle of at least four edges as an induced subgraph. The class of chordal graphs contains many famous graph classes such as trees, interval graphs, and split graphs, and is also a subclass of perfect graphs. In this paper, we address the problem of enumerating all labeled chordal graphs included in a given graph. We think of some variations of this problem. First we introduce an algorithm to enumerate all connected labeled chordal graphs in a complete graph of n vertices. Next, we extend the algorithm to an algorithm to enumerate all labeled chordal graphs in a n-vertices complete graph. Then, we show that we can use, with small changes, these algorithms to generate all (connected or not necessarily connected) labeled chordal graphs in arbitrary graph. All our algorithms are based on reverse search method, and time complexities to generate a chordal graph are O(1), and also O(1) delay. Additionally, we present an algorithm to generate every clique of a given chordal graph in constant time. Using these algorithms we obtain combinatorial Gray code like sequences for these graph structures in which the differences between two consecutive graphs are bounded by a constant size.