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.

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.

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.

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).

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.

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.

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 this paper, we propose and describe a new synchronizer for the FFT timing applicable to multi-carrier spread-spectrum (MC-SS) communication systems. The performance of the synchronizer is evaluated in terms of false- and miss-detection probabilities in the presence of additive white Gaussian noise (AWGN) and Rayleigh fading.

Frequency modulation (FM) noise spectroscopy with diode laser is applied to high-resolution Doppler-free spectroscopy of Cs atomic vapor near a dielectric surface with evanescent-wave pump-probe configuration. Both high resolution and high sensitivity are realized by using an extremely simple experimental setup, in which no sweep or precise tuning of laser frequency are required. Several experimental configurations of optical near-field spectroscopy are demonstrated, which is useful for an extensive study of resonant interactions of atoms and microscopic electronic systems in optical near-fields.

A performance comparison is developed between a chaotic communication system and a spread spectrum system with similar features in terms of bandwidth and transceiver structure but based on more conventional Gold sequences. Comparison is made in the presence of noise and multipath contributions which degrade the channel quality. It is shown that, because of its more favourable correlation properties, the chaotic scheme exhibits lower error rates, at a parity of the bandwidth expansion factor. The same favourable correlation properties are also used to explain and show, through a numerical example, the benefits of chaotic segments in a multi-user environment.

This paper presents the transmission performance of a class-IV partial-response signaling SSB system and proposes a method that can improve its power efficiency. A line code that has no dc component has been used in the SSB transmission of digital signals. The type of line code, such as a partial-response signaling, increases the modulation states, and as a result, decreases the power efficiency. To overcome this obstacle, a new demodulation method called "re-filtering and combining" is proposed on the assumption of orthogonal phase detection. The demodulated quadrature channel is re-filtered by a Hilbert filter and is combined with the in-phase channel. It is confirmed by computer simulations that the new demodulation method improves the BER performance and a 3 dB improvement of the power efficiency is obtained.

This paper addresses the single channel speech enhancement method which utilizes the mean value and variance of the logarithmic noise power spectra. An important issue for single channel speech enhancement algorithm is to determine the trade-off point for the spectral distortion and residual noise. Thus the accurate discrimination between speech spectral and noise components is required. The conventional methods determine the trade-off point using parameters obtained experimentally. As a result spectral discrimination is not adequate. And the enhanced speech is deteriorated by spectral distortion or residual noise. Therefore, a criteria to determine the point is necessary. The proposed method determines the trade-off point of spectral distortion and residual noise level by discrimination between speech spectral and noise components based on statistical criteria. The spectral discrimination is performed using hypothesis testing that utilizes means and variances of the logarithmic power spectra. The discriminated spectral components are divided into speech-dominant spectral components and noise-dominant ones. For the speech-dominant ones, spectral subtraction is performed to minimize the spectral distortion. For the noise-dominant ones, attenuation is performed to reduce the noise level. The performance of the method is confirmed in terms of waveform, spectrogram, noise reduction level and speech recognition task. As a result, the noise reduction level and speech recognition rate are improved so that the method reduces the musical noise effectively and improves the enhanced speech quality.

This paper presents a detailed simulation method to estimate Doppler power spectrum and mean blood velocity using real CW Doppler transducers with twin-crystal arrangement. The method is based on dividing the sample volume into small cells and using the statistics of the Doppler power spectrum with the same Doppler shift frequency, which predicts the mean blood velocity. The acoustic fields of semicircular transducers across blood vessels were calculated and the effects of acoustical and physiological factors on Doppler power spectrum and mean blood velocity were analyzed. Results show that nonuniformity of the acoustic field of the ultrasonic beam in the blood vessel and blood velocity profiles significantly affect Doppler power spectrum and mean blood velocity. However, Doppler angle, vessel depth, and sample volume length are not sensitive functions. Comparisons between simulation and experimental results illustrated a good agreement for parabolic flow profile. These results will contribute to a better understanding of Doppler power spectrum and mean blood velocity in medical ultrasound diagnostics.

An effective spatial resolution enhancement method for distributed strain measurement by BOTDR is proposed. An optical fiber is glued to a structure by a length less than the spatial resolution defined by the pulse width, and the Brillouin spectrum of the light scattered from the glued optical fiber is investigated theoretically. The apparent strain xp observed in the fiber is found to be proportional to the accurate strain a. The ratio r=xp/a coincides with the ratio of the glued length to the spatial resolution. Spatial resolution as small as 0.2 m is demonstrated experimentally for small strains of less than 10-3.

In this paper, we propose a novel higher order time-frequency distribution (GDH) for a discrete time signal. This distribution is defined over the original discrete time-frequency grids through a delicate discretization of an equivalent expression of a higher order distribution, for a continuous time signal, in [4]. We also present a constructive design method, for the kernel of the GDH, by which the distribution satisfies (i) the alias free condition as well as (ii) the marginal conditions. Numerical examples show that the proposed distributions reasonably suppress the artifacts which are observed severely in the Wigner distribution and its simple higher order generalization.