In this paper, we present a time-delay estimation algorithm in a closely spaced multipath environment for a direct-sequence code division multiple access (DS-CDMA) systems. The proposed scheme first converts the observed signal to the frequency domain by Fast Fourier Transform (FFT). We then formulate a nonlinear estimation problem, and convert it into a large scale linear least squares problem. We apply conjugate gradients iterations on the resulting normal equations to obtain the solution. Unlike the methods which invoke the criterion of nonlinear least squares, the proposed scheme can achieve higher resolution. The delay estimation is combined with a tracking algorithm based on a FIR prefilter. Simulation results confirm the effectiveness of the proposed algorithm in both AWGN and Rayleigh fading channels.
Chin-Jui LAI Ching-Her LEE Chung-I G. HSU Jean-Fu KIANG
A mode-matching technique in conjunction with the Floquet theorem is proposed to analyze the propagation characteristics of periodic circular surface waveguides. The circular waveguides are coated outside with a multilayered dielectric and have a ground plane with periodic corrugation of arbitrary profile. Three different ground corrugation profiles are examined to demonstrate the influences of the corrugation shape, depth, and width, dielectric thickness, and relative permittivity on bandstop characteristics.
In this letter, a hybrid selection combining (SC) and maximal ratio receive combining (MRRC) technique is proposed for orthogonal frequency-division multiplexing (OFDM) systems with multiple receive antennas. The proposed technique still uses multiple receive antennas, but it has just a single RF front-end and a single baseband demodulator. In comparison with the OFDM system with no diversity, we can achieve superior gain irrespective of bandwidth efficiency, and also in comparison with the MRRC OFDM, we can achieve better gain under the bandwidth efficiency of 3 bps/Hz at the bit error rate of 10-6.
The spatial distribution of the electric field in the low to high frequency bands radiated from printed circuit board (PCB) should be estimated continuously from near to far field. The characteristic of the electric field distribution is analyzed by the FDTD-multiple analysis space (FDTD-MAS) method, which can analyze from near to far field continuously, and compared with measured results. Since the analyzed electric field distribution is good agreement with measured results, it is suggested that the continuous distribution for electric field from near to far field can be calculated by the FDTD-MAS method. The electric field at low frequency is larger than that at high frequency within 1 m.
To estimate the number of substring matches against string data, count suffix trees (CS-tree) have been used as a kind of alphanumeric histograms. Although the trees are useful for substring count estimation in short data strings (e.g. name or title), they reveal several drawbacks when the target is changed to extremely long strings. First, it becomes too hard or at least slow to build CS-trees, because their origin, the suffix tree, has memory-bottleneck problem with long strings. Secondly, some of CS-tree-node counts are incorrect due to frequent pruning of nodes. Therefore, we propose the count q-gram tree (CQ-tree) as an alphanumeric histogram for long strings. By adopting q-grams (or length-q substrings), CQ-trees can be created fast and correctly within small available memory. Furthermore, we mathematically provide the lower and upper bounds that the count estimation can reach to. To the best of our knowledge, our work is the first one to present such bounds among research activities to estimate the alphanumeric selectivity. Our experimental study shows that the CQ-tree outperforms the CS-tree in terms of the building time and accuracy.
Junni ZOU Hongkai XIONG Rujian LIN
To simultaneously support guaranteed real-time services and best-effort service, a Priority-based Scheduling Architecture (PSA) designed for high-speed switches is proposed. PSA divides packet scheduling into high-priority phase and low-priority phase. In the high-priority phase, an improved sorted-priority algorithm is presented. It introduces a new constraint into the scheduling discipline to overcome bandwidth preemption. Meanwhile, the virtual time function with a control factor α is employed. Both computer simulation results and theoretic analysis show that the PSA mechanism has excellent performance in terms of the implementation complexity, fairness and delay properties.
Masahiro NOMURA Taku OHSAWA Koichi TAKEDA Yoetsu NAKAZAWA Yoshinori HIROTA Yasuhiko HAGIHARA Naoki NISHI
This paper describes a newly developed automatic direction control scheme for bi-directional bus repeaters that uses dynamic collaborative driving techniques. Repeater directions are rapidly determined by detecting the direction of control signal propagation through an additional control signal line that is driven by dynamic collaborative drivers. Application to an on-chip peripheral bus reduces control circuit transistor counts by about 75% and the number of control signal lines by about 50% without loss of speed. Experimental results for a 0.18-µm CMOS implementation indicate that the proposed scheme is four times faster than a conventional scheme with no bi-directional bus repeaters.
Masakiyo FUJIMOTO Satoshi NAKAMURA
This paper addresses a speech recognition problem in non-stationary noise environments: the estimation of noise sequences. To solve this problem, we present a particle filter-based sequential noise estimation method for front-end processing of speech recognition in noise. In the proposed method, a noise sequence is estimated in three stages: a sequential importance sampling step, a residual resampling step, and finally a Markov chain Monte Carlo step with Metropolis-Hastings sampling. The estimated noise sequence is used in the MMSE-based clean speech estimation. We also introduce Polyak averaging and feedback into a state transition process for particle filtering. In the evaluation results, we observed that the proposed method improves speech recognition accuracy in the results of non-stationary noise environments a noise compensation method with stationary noise assumptions.
Kenjiro MATSUOKA Kazushi SAEKI Eiji TERAOKA Minoru YAMADA Yuji KUWAMURA
Properties of the quantum noise and the optical feedback noise in blue-violet InGaN semiconductor lasers were measured in detail. We confirmed that the quantum noise in the blue-violet laser becomes higher than that in the near-infrared laser. This property is an intrinsic property basing on principle of the quantum mechanics, and is severe subject to apply the laser for optical disk with the small consuming power. The feedback noise was classified into two types of "low frequency type" and "flat type" basing on frequency spectrum of the noise. This classification was the same as that in the near infra-red lasers.
Gabriel Porto VILLARDI Giuseppe Thadeu Freitas de ABREU Ryuji KOHNO
The application of Orthogonal Space-Time Block Codes (O-STBC) as the encoding scheme in the presence of "non-quasi-static" fading was considered. A simple and efficient adaptive method of channel estimation based on the interpolation of estimates acquired at the pre-amble and post-amble of framed blocks of information is developed. Moreover, the proposed method is proven, both theoretically and by simulations, to outperform the alternative of channel tracking, despite its significant low complexity.
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.
Yi QIAN Rose Qingyang HU Catherine ROSENBERG
There are many system proposals for satellite-based broadband communications that promise high capacity and ease of access. Many of these proposals require advanced switching technology and signal processing on-board the satellite(s). One solution is based on a geo-synchronous (GEO) satellite system equipped with on-board processing and on-board switching. An important feature of this system is allowing for a maximum number of simultaneous users, hence, requiring effective medium access control (MAC) layer protocols for connection admission control (CAC) and bandwidth on demand (BoD) algorithms. In this paper, an integrated CAC and BoD algorithm is proposed for a broadband satellite communication system with heterogeneous traffic. A detailed modeling and simulation approach is presented for performance evaluation of the integrated CAC and BoD algorithm based on heterogeneous traffic types. The proposed CAC and BoD scheme is shown to be able to efficiently utilize available bandwidth and to gain high throughput, and also to maintain good Grade of Service (GoS) for all the traffic types. The end-to-end delay for real-time traffic in the system falls well within ITU's Quality of Service (QoS) specification for GEO-based satellite systems.
Seokho YOON Suk Chan KIM Sun Yong KIM
Recently, a novel detector was proposed by the authors for code acquisition in non-Gaussian impulsive channels [3], which dramatically outperforms the conventional squared-sum detector; however, it requires exact knowledge of the non-Gaussian noise dispersion. In this paper, a robust detector is proposed, which employs the signs and ranks of the received signal samples, instead of their actual values, and so does not require knowledge of the non-Gaussian noise dispersion. The acquisition performance of the proposed detector is compared with that of the detector of [3] in terms of the mean acquisition time. The simulation results show that the proposed scheme is not only robust to deviations from the true value of the non-Gaussian noise dispersion, but also has comparable performance to that of the scheme of [3] using exact knowledge of the non-Gaussian noise dispersion.
Yasuo SATO Shuji HAMADA Toshiyuki MAEDA Atsuo TAKATORI Seiji KAJIHARA
In this paper we introduce a statistical quality model for delay testing that reflects fabrication process quality, design delay margin, and test timing accuracy. The model provides a measure that predicts the chip defect level that cause delay failure, including marginal small delay. We can therefore use the model to make test vectors that are effective in terms of both testing cost and chip quality. The results of experiments using ISCAS89 benchmark data and some large industrial design data reflect various characteristics of our statistical delay quality model.
Hiroshi YOSHIOKA Yushi SHIRATO Kazuji WATANABE
We propose a novel simplified Viterbi equalizer for high symbol rate FWA (Fixed Wireless Access) systems carrying 64QAM signals. Reduced complexity and improved performance are achieved adopting two approaches. The first one is reducing the number of survival paths, taking advantage of the large D/U common in LOS (line of sight) communications. The second one is using a multi-stage process to generate desired signal replicas based on their likelihoods. Computer simulations confirm that the proposed replica generation method offers a performance improvement of about 1 dB and the proposed Viterbi equalizer offers reduced complexity with no performance penalty compared to full Viterbi equalizer.
M. Shahidur RAHMAN Tetsuya SHIMAMURA
A new system identification based method has been proposed for accurate estimation of vocal tract parameters. An often encountered problem in using the conventional linear prediction analysis is due to the harmonic structure of the excitation source of voiced speech. This harmonic characteristic is coupled with the estimation of autoregressive (AR) coefficients that results in difficulties in estimating the vocal tract filter. This paper models the effective voice source from the residual obtained through the covariance analysis in the first-pass which is then used as input to the second-pass least-square analysis. A better source-filter separation is thus achieved. The formant frequencies and corresponding bandwidths obtained using the proposed method for synthetic vowels are found to be accurate up to a factor of more than three (in percent) compared to the conventional method. Since the source characteristic is taken into account, local variations due to the positioning of analysis window are reduced significantly. The validity of the proposed method is also examined by inspecting the spectra obtained from natural vowel sounds uttered by high-pitched female speaker.
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.
Feng LIU Shaoqian LI Min LIANG Laizhao HU
A new wideband signal DOA estimation algorithm based on modified quantum genetic algorithm (MQGA) is proposed in the presence of the errors and the mutual coupling between array elements. In the algorithm, the narrowband signal subspace fitting method is generalized to wideband signal DOA finding according to the character of space spectrum of wideband signal, and so the rule function is constructed. Then, the solutions is encoded onto chromosomes as a string of binary sequence, the variable quantum rotation angle is defined according to the distribution of optimization solutions. Finally, we use the MQGA algorithm to solve the nonlinear global azimuths optimization problem, and get optimization azimuths by fitness values. The computer simulation results illustrated that the new algorithm have good estimation performance.
Yoshiaki YAMADA Satoru OHTA Hitoshi UEMATSU
Time synchronization is indispensable for wide area distributed systems including sensor networks, automation systems, and measurement/control systems. Another application is clock distribution, which is indispensable to support continuous information transfer. Because of the increasing demand for more sophisticated applications, it is essential to establish a time synchronization technique that offers higher accuracy and reliability. Particularly, the accuracy of time synchronization for Ethernet must be enhanced since Ethernet is becoming more important in telecommunication networks. This paper investigates a precise time synchronization technique that supports Gb/s Ethernet. To obtain accurate time synchronization, delay variation in message transfer and processing must be minimized. For this purpose, the paper first describes the implementation of preemptive priority queuing, which decreases the message delay variation of Ethernet. Through experiments, it is shown that preemptive priority queuing effectively achieves very low delay variation. The paper then proposes a method to synchronize the time signal of a slave node to that of the master node. The proposed time synchronization method is performed in the lower protocol layer and implemented on FPGA-based hardware. The method achieves superior time accuracy through the low message transfer/processing delay variation provided by preemptive priority, lower layer execution, and hardware implementation. The effectiveness of the method is confirmed through experiments. The experiments show that the time variation achieved by the method is smaller than 0.1 µsec. This performance is better than those obtained by existing synchronization methods.
Tetsuki TANIGUCHI Hoang Huy PHAM Nam Xuan TRAN Yoshio KARASAWA
This paper presents a simple method to determine weights of single carrier multiple input multiple output (MIMO) broadband communication systems adopting tapped delay line (TDL) structure in receiver side for the effective communication under frequency selective fading (FSF) environment. First, assuming the perfect knowledge of the channel matrix in both arrays, an iterative design method of transmitter and receiver weights is proposed. In this approach, both weights are determined alternately to maximize signal to noise plus interference ratio (SINR) by fixing the weight of one side while optimizing the other, and this operation is repeated until SINR converges. Next, considering the case of uninformed transmitter, maximum SINR design method of MIMO system is extended for space time block coding (STBC) scheme working under FSF. Through computer simulations, it is demonstrated that the proposed schemes achieves higher SINR than conventional method with delay-less structure, particularly for the fading with long duration.