A new technique for optical generation of high-purity millimeter-wave (mm-wave) signals--namely, by synthesizing the outputs from cascadingly phase-locked multiple semiconductor lasers--was developed. Firstly, a high-spectral-purity mm-wave signal was optically generated by heterodyning the outputs from two phase-locked external-cavity semiconductor lasers. The beat signal was detected by a p-i-n photodiode whose output was directly coupled to a coax-waveguide converter followed by a W-band harmonic mixer. By constructing an optical phase-locked loop (OPLL), a high-spectral-purity mm-wave signal with an electrical power of 2.3 µW was successfully generated at 110 GHz with an rms phase fluctuation of 57 mrad. Secondly, the frequency of the mm-wave signal was extended by use of three cascadingly phase-locked semiconductor lasers. This technique uses a semiconductor optical amplifier (SOA) to generate four-wave-mixing (FWM) signals as well as to amplify the input signals. When the three lasers were appropriately tuned, two pairs of FWM signals were nearly degenerated. By phase-locking the offset frequency in one of the nearly degenerated pairs, the frequency separations among the three lasers were kept at a ratio of 1:2. Thus, we successfully generated high-purity millimeter-wave optical-beat signals at frequencies at 330.566 GHz with an rms phase fluctuation of 0.38 rad. A detailed analysis of the phase fluctuations was carried out on the basis of measured power spectral densities. The possibility of extending the mm-wave frequency up to 1 THz by using four cascadingly phase-locked lasers was also discussed.

Design patterns can be regarded as an approach to encapsulate and reuse good design practices. However, most design patterns are specified using informal text and examples. To obtain all of the benefits of patterns, formal specification and tool support are indispensable. This paper proposes a Design Pattern Specification Language (DPSL) that is both manageable and effective. The DPSL provides software developers with the capability to treat design patterns as concrete design units without lowering abstraction. To demonstrate the usability of our DPSL and its application in design modeling, we have developed a prototype tool that supports the DPSL in UML diagrams. This prototype allows us to demonstrate the tool's support possibilities and the usability of patterns for software development applications.

In order to establish rapid diagnosing in TDX signaling service, it has developed a PCM signal acquisition (PCMA) system which can analyze status of signals sent from/received to a signaling equipment, providing the fully electronic switching system. The system has a function of acquirement PCM signal of the preferred channel from the subhighway (SHW), connecting a universal signal transceiver unit (USTU) and time switch unit (TSU), and then it classifies the type of signal such as R2MFC/DTMF/CCT/VOICE, and finally discriminates the digit. This paper analyze the signal status of the PCMA system using the quick Fourier transform (QFT) based the symmetric properties, and discusses the algorithm of signal analysis and discrimination. In the experimental results, it shows the improved performance to the PCMA and reduce memory waste and process the real-time.

This paper proposes a new multispectral image data compression algorithm that can efficiently reduce spatial and spectral redundancies by applying classified prediction, a Karhunen-Loeve transform (KLT), and the three-dimensional set partitioning in hierarchical trees (3-D SPIHT) algorithm in the wavelet transform (WT) domain. The classification is performed in the WT domain to exploit the interband classified dependency, while the resulting class information is used for the interband prediction. The residual image data on the prediction errors between the original image data and the predicted image data is decorrelated by a KLT. Finally, the 3-D SPIHT algorithm is used to encode the transformed coefficients listed in a descending order spatially and spectrally as a result of the WT and KLT. Simulation results showed that the reconstructed images after using the proposed algorithm exhibited a better quality and higher compression ratio than those using conventional algorithms.

In this paper, to enhance the power efficiency a new simple space-time coding scheme is devised with application to the OFDM based Wireless LAN system. The basic idea is from the receiver's point of view and is referred to as Virtual Constellation Mapping (VCM). We designed a new combination of the channel coding (Turbo Code) along with multiple transmit antennas (Two antennas) to achieve transmit diversity and space division multiplexing transmission. Computer simulation results showed that with the same transmission data rate, our proposed scheme can achieve better bit error rate (BER) compared with the conventional space-time trellis coded OFDM scheme in high Doppler fading channels.

The main contribution of this paper is to present an image retrieval system using FPGAs. Given a template image T and a database of a number of images I1, I2,, our system lists all images that contain a subimage similar to T. More specifically, a hardware generator in our system creates the Verilog HDL source of a hardware that determines whether Ii has a similar subimage to T for any image Ii and a particular template T. The created Verilog HDL source is compiled and embedded in an FPGA using the design tool provided by the FPGA vendor. Since the hardware embedded in the FPGA is designed for a particular template T, it is an instance-specific hardware that allows us to achieve extreme acceleration. We evaluate the performance of our image matching hardware using a PCI-connected Xilinx FPGA and a timing analyzer. Since the generated hardware attains up to 3000 speed-up factor over the software solution, our approach is promising.

It is found that the supercontinuum spectrum is generated from cross-phase modulated soliton pulses which are propagated through a dispersion-flattened/decreasing fiber with low birefringence. The cross-phase modulation is achieved by exciting two orthogonally polarized modes in a birefringent fiber and the effect of input azimuth of linearly polarized pulses is discussed theoretically and numerically.

A new analog correlator circuit is proposed for direct sequence code division multiple access (DS-CDMA) demodulator. The circuit consists of only 16 switches, 4 capacitors and 2 level shifters. Control sequence requires only three clock phases. Simulation with code length of 127 reveals that the proposed circuit has a good ability to cancel off the charge error and dissipates 3.4mW at 128MHz. The circuit had been designed using a 0.6µm CMOS process. The area of 256µm 245µm is estimated to be 9 times smaller compared to other reported equivalent analog correlators.

An automatic LSI package lead inspection system for backside lead specification is proposed. The proposed system inspects not only lead backside contamination but also the mechanical lead specification such as lead pitch, lead offset and lead overhangs (variations in lead lengths). The total inspection time of a UQFP package with a lead count of 256 is less than the required time of 1 second. Our proposed method is superior to the threshold method used usually, especially for the defect between leads.

This paper proposes a low power consumption Analog Matched Filter (AMF) that utilizes capacitor multiply-and-accumulate operations. A high-speed, high-precision Analog-to-Digital (A/D) converter is unnecessary because the proposed circuit directly samples received analog signals. A code-shifting MF structure is used to prevent errors from accumulating. A 15-tap AMF circuit was fabricated using 0.35 µm CMOS technology. Power consumption for the 128-tap circuit is estimated to be 22.3 mW at 25 MHz and 3.3 V, and the area is estimated to be 0.33 mm2. The proposed circuit will thus be a useful LSI for mobile terminals.

GPS receivers are susceptible to jamming by interference. This paper proposes a recurrent neural network (RNN) predictor for new application in GPS anti-jamming systems. Five types of narrowband jammers, i. e. AR process, continuous wave interference (CWI), multi-tone CWI, swept CWI, and pulsed CWI, are considered in order to emulate realistic conditions. As the observation noise of received signals is highly non-Gaussian, an RNN estimator with a nonlinear structure is employed to accurately predict the narrowband signals based on a real-time learning method. The node decoupled extended Kalman filter (NDEKF) algorithm is adopted to achieve better performance in terms of convergence rate and quality of solution while requiring less computation time and memory. We analyze the computational complexity and memory requirements of the NDEKF approach and compare them to the global extended Kalman filter (GEKF) training paradigm. Simulation results show that our proposed scheme achieves a superior performance to conventional linear/nonlinear predictors in terms of SNR improvement and mean squared prediction error (MSPE) while providing inherent protection against a broad class of interference environments.

In this paper, we present a novel beam-former capable of tracking a rapidly moving speaker in a very noisy environment. The localization algorithm extracts a set of candidate direction-of-arrival (DOA) for the signal sources using array signal processing methods in the frequency domain. A minimum variance (MV) beam-former identifies the speech signal DOA in the direction where the signal's spectrum entropy is minimized. A fine tuning process detects the MV direction which is closest to the initial estimation using a smaller analysis window. Extended experiments, carried out in the range of 20-0 dB SNR, show significant improvement in the recognition rate of a moving speaker especially in very low SNRs (from 11.11% to 43.79% at 0 dB SNR in anechoic environment and from 9.9% to 30.51% in reverberant environment).

In this paper, we propose a new technique of filter bank subtraction for robust speech recognition under various acoustic conditions. Spectral subtraction is a simple and useful technique for reducing the influence of additive noise. Conventional spectral subtraction assumes accurate estimation of the noise spectrum and no correlation between speech and noise. Those assumptions, however, are rarely satisfied in reality, leading to the degradation of speech recognition accuracy. Moreover, the recognition improvement attained by conventional methods is slight when the input SNR changes sharply. We propose a new method in which the output values of filter banks are used for noise estimation and subtraction. By estimating noise at each filter bank, instead of at each frequency point, the method alleviates the necessity for precise estimation of noise. We also take into consideration expected phase differences between the spectra of speech and noise in the subtraction and control a subtraction coefficient theoretically. Recognition experiments on test sets at several SNRs showed that the filter bank subtraction technique improved the word accuracy significantly and got better results than conventional spectral subtraction on all the test sets. In other experiments, on recognizing speech from TV news field reports with environmental noise, the proposed subtraction method yielded better results than the conventional method.

In this paper, a grey filtering approach based on GM(1,1) model is proposed. Then the grey filtering is applied to speech enhancement. The fundamental idea in the proposed grey filtering is to relate estimation error of GM(1,1) model to additive noise. The simulation results indicate that the additive noise can be estimated accurately by the proposed grey filtering approach with an appropriate scaling factor. Note that the spectral subtraction approach to speech enhancement is heavily dependent on the accuracy of statistics of additive noise and that the grey filtering is able to estimate additive noise appropriately. A magnitude spectral subtraction (MSS) approach for speech enhancement is proposed where the mechanism to determine the non-speech and speech portions is not required. Two examples are provided to justify the proposed MSS approach based on grey filtering. The simulation results show that the objective of speech enhancement has been achieved by the proposed MSS approach. Besides, the proposed MSS approach is compared with HFR-based approach in [4] and ZP approach in [5]. Simulation results indicate that in most of cases HFR-based and ZP approaches outperform the proposed MSS approach in SNRimp. However, the proposed MSS approach has better subjective listening quality than HFR-based and ZP approaches.

It is believed that distant-talking speech recognition in a noisy environment requires a large-scale microphone array. However, this cannot fit into small consumer devices. Our objective is to improve the performance with a limited number of microphones (preferably only left and right). In this paper, we focused on a profile that is the shape of the power distribution according to the beamforming direction. An observed profile can be decomposed into known profiles for directional sound sources and a non-directional background sound source. Evaluations confirmed this method reduced the CER (Character Error Ratio) for the dictation task by more than 20% compared to a conventional 2-channel Adaptive Spectral Subtraction beamformer in a non-reverberant environment.

This paper describes a simple, inexpensive technique for intentionally broadening and flattening the spectrum of a DC-DC converter (switching regulator) to reduce Electro-Magnetic Interference (EMI). This noise spectrum broadening technique involves intentionally introducing pseudo-random dithering of control clock timing, which can be achieved by adding simple digital circuitry. This technique can significantly reduce noise power spectrum peaks at the DC-DC converter output. For our test case circuit, measurements showed that noise power was reduced by 5.7 dBm at the main peak, by 15.6 dBm at the second peak and by 12.8 dBm at the third peak. This simple, inexpensive technique can be applied to most conventional switching regulators by adding simple digital circuitry, and without any modification of the design of other parts.

In wireless communication systems, low-power metrics is becoming a burdensome problem in the portable terminal design, because of portability constraints. This paper presents design architecture of a low-power Digital Matched Filter (DMF) for the direct-sequence spread-spectrum communication system such as WCDMA or wireless LAN. The proposed approach for power savings focuses on the architecture of the reception registers and the correlation-calculating unit, which dissipate the majority of the power in a DMF. The main features are asynchronous latch clock generation for the reception registers, parallelism of correlation calculation operations and bit manipulation for chip-correlation operations. A DMF is designed in compliance with the WCDMA specifications incorporating the proposed techniques, and its properties are evaluated by computer simulations at the gate level using 0.18-µm CMOS standard cell array technology. As a result, the power consumption of the proposed DMF is estimated to be 9.3 mW (@15.6 MHz, 1.6 V), which is below 40% of the power consumed by a general DMF.

This paper discusses two types of polyphase sequence sets, which will successfully provide CDMA systems without co-channel interference. One is a type of ZCZ sets, whose periodic auto-correlation functions take zero at continuous shifts on both side of the zero-shift, and periodic cross-ones also take zero at the continuous shifts and the zero-shift. The other is a new type of sets consisting of some subsets of polyphase sequences with zero cross-correlation zone, called ZCCZ sets, whose periodic cross-correlation functions among different subsets have take zero at continuous shifts on both side of the zero-shift including the zero-shift. The former can achieve a mathematical bound, and the latter can have large size.

Many types of adaptive algorithms based on the MMSE criterion for co-channel interference suppression in DS/CDMA systems have been studied in great detail. However, these algorithms have such a problem that the training speed is greatly dropped under the strong near-far problem. In this paper, we propose and analyze an adaptive filter based on the Maximum Signal to Interference and Noise Ratio (MSINR) criterion, called adaptive MSINR filter. This filter is basically equivalent to the adaptive filter based on the MMSE criterion. However, due to the structual difference, the convergence speed is greatly improved. Specifically, the de-spreading vector in this filter is so renewed as to maximize the Signal to Interference and Noise Ratio (SINR) by minimizing the de-spread interference and noise power under the condition that the de-spread desired signal power keeps constant. So the proposed filter uses the estimated interference and noise signal calculated by subtracting the estimated desired signal from the received signal. It is just the reason why the adaptive MSINR filter shows remarkable convergence speed. And to satisfy the constant signal power condition, the projection matrix onto the orthogonal complement of the desired signal space is used for the de-spreading vector. For the proposed filter, we analyze the convergence modes and also investigate the de-spread interfernce and noise power for calculating the theoretical SINR curve. Then, we conduct some computer simulations in order to show the difference between this filter and the conventional one in terms of the SINR convergence speed. As the result, we confirm that the adaptive filter based on the MSINR criterion achieves significant progress in terms of the SINR convergence speed.

In an order-specified multisignature scheme, one can verify not only a set of signers who have signed the message but also its signing order. Though we have seen several schemes with such properties proposed, none of them is given the security proof against active adversaries. The scheme can be easily modified to be an order-specified multisignature scheme, but still has the restriction that the possible signing orders are only ones of the type of serial signing. In this paper, we propose the first order-specified multisignature scheme, which is shown to be secure against adaptive chosen-message insider attacks in the random oracle model, and which allows the signing orders to form like any series-parallel graphs. The security is shown by using ID-reduction technique, which reduces the security of multisignature schemes to those of multi-round identification schemes. Furthermore, we discuss the efficiency of the proposed scheme and the upper bound of the possible number of participating signers.