This letter propose a fast frequency domain adaptive filter algorithm (FADF) for applications in which large order adaptive filters are required. Proposed FADF algorithm reduces the block delay of conventional FADF algorithms allowing a more efficient selection of the fast Fourier Transform (FFT) size. Proposed FADF algorithm also provides faster convergence rates than conventional FBAF algorithms by using a near-optimum convergence factor derived by using the FFT. Computer simulations using white and colored signals are given to show the desirable features of proposed scheme.

We have developed the Epi-Base SATURN process as a silicon bipolar process technology which can be applied to optical transmission LSIs. This process technology, to which low temperature selective epitaxial growth technology is applied, is based on the SATURN process. By performing selective epitaxial growth for base formation in 2 steps, transistors with a 40GHz maximum cut-off frequency have been fabricated. In circuit simulation based on SPICE parameters of transistors, the target performance required for 2.4 Gbit/s optical interface LSIs has been achieved.

A new superresolution technique is proposed for high-resolution estimation of the scattering analysis. For complicated multipath propagation environment, it is not enough to estimate only the delay-times of the signals. Some other information should be required to identify the signal path. The proposed method can estimate the frequency characteristic of each signal in addition to its delay-time. One method called modified (Root) MUSIC algorithm is known as a technique that can treat both of the parameters (frequency characteristic and delay-time). However, the method is based on some approximations in the signal decorrelation, that sometimes make problems. Therefore, further modification should be needed to apply the method to the complicated scattering analysis. In this paper, we propose to apply a time-domain null filtering scheme to reduce some of the dominant signal components. It can be shown by a simple experiment that the new technique can enhance estimation accuracy of the frequency characteristic in the Root-MUSIC algorithm.

An adaptive decoding scheme for a concatenated code used in the frequency-hopped spread-spectrum communication system in the presence of a pulse-burst jammer is proposed and its performance is analyzed. Concatenated coding schemes employing binary inner-code and Reed-Solomon outer code are investigated and the use of side information is allowed to decode both erasures and errors. The proposed scheme makes the decoder enable to adapt to the jamming level by switching between two decoding modes such that the decoded bit error rate can be reduced. The optimal threshold value for switching in this proposed scheme is derived. It has been shown that the proposed decoding scheme yields a significant performance improvement over a conventional decoding scheme. In addition, performance analysis and its variation of adaptive decoding scheme with the imperfect side information are also presented.

This paper proposes a new digitized group modulator and demodulator (a group modem) for adaptive frequency hopping and multi-carrier (AFHMC) radio systems. The group modem can flexibly vary the number of carriers handled simultaneously, especially employing a time division multiplexing technique in the demodulator. We discuss the operational principle of the modem. The required operational clock frequency in the group demodulator is also examined and clarified taking into consideration the frequency characteristics of the baseband filter. The basic performance of the proposed configuration is measured experimentally by constructing a π/4-shift QPSK group modulator and a π/4-shift QPSK group demodulator. First, by measuring the output spectrum of the significant parts in the demodulator, we confirm that the basic operational performance conforms to the design specifications. Secondly, investigating the relationship between the number of multiplexed low-pass filter taps and the required CNR when multiple carriers are simultaneously input confirms that more than 40 taps are enough to obtain the best BER performance in this experiment. Next, examining the relationship between the number of carriers simultaneously input, the required CNR, and the input level of these carriers confirm that the required CNR is roughly constant and there is no significant difference among the cases when D/U is more than 0 dB. Finally, an experiment shows that the required number of quantization bits for A/D input in the demodulator is more than 6, which is enough to obtain the best BER even if simultaneous handled carriers are 4.

This letter presents a new design method for approximate inverse systems using all-pass networks. The efficacy of approximate inverse systems for input and parameter estimation of nonminimum phase systems is well recognized. in the previous methods, only time domain design of FIR (finite impulse response) type approximate inverse systems were considered. Here, we demonstrate that IIR (infinite impulse response) type approximate inverse systems outperform the previous methods. A nonlinear optimization technique is adopted for designing the proposed system in the frequency domain. Numerical examples are also presented to show the effectiveness of the proposed method.

In general, a time-limited signal such as a single sinusoidal waveform framed by a frame period T can be utilized for conveying a multi-level symbol in data transmission. If such a signal is analyzed by the conventional DFT (Discrete Fourier Transform) analysis, the infinite number of frequency components with frequency spacing fD = T1 is needed. This limits the accuracy with which the original frequency of the unframed sinusoidal waverform can be identified. It is especially difficult to identify two similar framed sinusoids whose frequency spacing is narrower than fD. An analytical principle for time-limited signals is therefore proposed by introducing the concept of an Extended Frame into DFT. Waveform analysis more accurate than DFT is achieved by taking into account multiple correlations between extended frames made of an input frame signal and the element frequency components corresponding to the length of each extended frame. In this approach, it is possible to use arbitrary element frequency spacing less than fD. It also allows an element frequency to be selected as a real number times of fD, rather than as an integer times of fD that is used for DFT. With this analyzing mechanism, it is verified that an input frame signal with only the frequency components which coincide with any of the element frequencies can be exactly analyzed. The disturbance caused by the input white noise is examined. As a result, it is found that the superior noise suppression function is achieved by this method over a conventional matched filter. In addition, the error caused by using a finite number of element frequencies and the A/D conversion accuracy required for sampling an input signal are examined, and it is shown that these factors need not impede practical implementation. For this reason, this principle is useful for multi-ary transmission systems, noise tolerant receivers, or systems requiring precise filtering of time limited waveforms.

According as the fine LSI process technique develops, the technique to reduce power dissipation of high-frequency integrated analog circuits is getting more important. This paper describes a design of high-frequency integrator with low power dissipation for monolithic leapfrog filters. In the design of the conventional monolithic integrators, there has been a great dfficulty that a high-frequency integrator which can operate at low supply voltage cannot be realized without additional circuits, such as unbalanced-to-balanced conversion circuits and common-mode feedback circuits. The proposed integrator is based on the Miller integrator. By a PNP current mirror circuit, high CMRR is realized. However, the high-frequency characteristic of the integrator is independent of PNP transistors. In addition, it can operate at low supply voltage. The excess phase shift of the integrator is compensated by insertion of the compensation capacitance. The effectiveness of the proposed technique is confirmed by PSPICE simulation. The simulation results of the integrator shows that the common-mode gain is efficiently low and the virtual ground is realized, and that moderate phase compensation can be achieved. The simulation results of the 3rd-order leapfrog filter using the integrator shows that the 50 MHz-cutoff frequency filter is obtained. Its power dissipation in operating 2 V-supply voltage is 5.22 mW.

A new configuration is proposed for an optoelectronic network (OEN) using microwave frequency mixing and multiplexing. The mn OEN consists of m optical sources, m-parallel n-stage cascaded optical intensity modulators, and m-photodetectors. The mn OEN matrix is theoretically discussed, and 12, 22 and 33 OENs are analyzed in detail. The 22 OEN, which mixes and multiplexes microwaves, is further investigated and the theoretical prediction derived from OEN equations is experimentally confirmed.

In order to evaluate quantitatively TMJ sound, TMJ sound in normal subject group, CMD patient group A with palpable sounds unknown to them, CMD patient group B with palpable sounds known to them, and CMD patient group C with audible sounds were detected by a contact microphone, and frequency analysis of the power spectra was performed. The power spectra of TMJ sound of normal subject group and patient group A showed patterns with frequency values below 100 Hz, whereas the power spectra of patient groups B and C showed distinctively different patterns with peaks of frequency component exceeding 100 Hz. As regards the cumulative frequency value, the patterns for each group clearly differed from those of other groups; in particular the 80% cumulative frequency value showed the greatest difference. From these results, it is assumed that the 80% cumulative frequency value can be used as an effective indicator for quantitative evaluation of TMJ sound.

In this paper, electromagnetic scattering by infinite double two-dimensional periodic array of resistive upper and lower elements is considered. The electric field equations are solved by using the moment method in the spectral domain. Some numerical results are shown and frequency selective properties are discussed.

In this work, a statistical analysis is performed for a simple constrained high-order Yule-Walker (YW) tone frequency estimator obtained from the first equation of the constrained high-order YW equations. Explicit expressions for its estimation bias and variance are efficiently derived by virtue of a Taylor series expansion technique. Especially, being explicit in terms of frequency, data length and Signal-to-Noise Ratio (SNR) value, the resulting bias expression can not be obtained by using the asymptotic analyses used for the parameter estimation methods. The obtained expressions are compared with their counterparts of the Pisarenko tone frequency estimator. Simulations are performed to support the theoretical results.

Frequency demodulation of a single side-band signal with a carrier is discussed and a new demodulation method is proposed. Compared with the conventional RZSSB (real zero single-side band) demodulator, there are no even-order distortions in the demodulated output signal of the proposed method and the third-order distortion can be canceled very easily without any Hilbert transformer as is required in the conventional RZSSB demodulator. Since the carrier can be reproduced completely from the input signal by the proposed method, it can be used for not only an RZSSB signal but also a full-carrier SSB signal or a low-carrier SSB signal. Compared with transmitting an RZSSB signal, it more efficiently transmits a low carrier SSB signal since the carrier does not include any useful information. By this means, the transmission efficiency can be greatly improved by adopting the proposed method.

Considered is the theory of several dielectric waveguide phenomena for which the vector nature of the electromagnetic field is essential. These phenomena are the following rotation of the plane of polarization in chiral and twisted waveguides, Bragg's reflection in a twisted waveguide in a narrow frequency band, and excitation of a waveguide at a near-cutoff frequency.

This paper proposes a multiple optical wideband frequency modulation system and clarifies its phase noise insensitivity. In this system, an optical carrier is phase-modulated by a conventional FM signal to generate many sidebands in optical frequency band. The n-th order sideband component yields also FM signal with frequency deviation of n times the one of original FM signal. Therefore, by selecting the high order optical sideband, the wideband optical FM signal can be obtained. Moreover, if some sidebands are simultaneously extracted and multiplied at the receiver, a wideband FM signal with larger frequency deviation and no laser phase noise can be obtained, and FM threshold extension can be realized.

This paper discusses the performance of non-coherent reception for M-ary spread-spectrum (M-ary/SS) signals in the presence of carrier frequency offset. In general, the M-ary/SS scheme is expected to be of higher spectral efficiency than the conventional DS/SS schemes, but its performance may be degraded by the carrier frequency offset. We, therefore, analyze the effect of carrier frequency offset on the performance of the non-coherent M-ary/SS system with orthogonal modulation using a set of sequences generated by the Hadamard matrix. As a result of the analysis, it has been found that the carrier frequency offset may cause a great deal of degradation in the performance, and that its effect has a distinctive property which is due to the characteristic of Hadamard matrix, at the same time. Making use of this property, we propose two schemes that can mitigate the effect of carrier frequency offset: one is based on choise of the code sequences, the other is on the error correcting code. The effectiveness of the schemes is evaluated in the terms of symbol-error-rates through analysis and computer simulation.

Time-frequency representations (TFRs) have been developed as tools for analysis of non-stationary signals. Signal dependent TFRs are known to perform well for a much wider range of signals than any fixed (signal independent) TFR. This paper describes customised and sequential versions of the signal dependent TFR proposed in [1]. The method, which is based on the use of the Radon transform at distance zero in the ambiguity domain, is simple and effective in dealing with both simulated and real data. The use of the described method for time-scale analysis is also presented. In addition, the paper investigates a simple technique for detection of noisy chirp signals using the Radon transfrom in the ambiguity domain.

A 0.1-µm gate-length GaAs MESFET technology is reported. A 48.3-GHz dynamic-frequency divider, and an amplifier with 20-dB gain and 17.5-GHz bandwidth are successfully fabricated by integrating over-100-GHz-cut-off frequency MESFETs using a new lightly-doped drain structure with a buried p-layer (BP-LDD) device structure.

This paper proposes a technique for estimating the harmonic frequencies based on instantaneous frequency (IF) of speech signals. The main problem is how to decompose the speech signal into the harmonic components. For this purpose, we use a set of bandpass-filters, each of whose center frequencies changes with time in order to track the instantaneous freuency of its output. As a result, the outputs of the band-pass filters become the harmonic components, and the instantaneous frequencies of the harmonics are accurately estimated. To evaluate the effectiveness of the approach, we apply it to pitch determination of speech. Pitch determination is simply accomplished by selecting the correct fundamental frequency out of the harmonic components. It is confirmed that the pitch extraction using the proposed pitch determination algorithm (PDA) is stable and accurate. The most significant feature of the PDA is that the extracted pitch contour is smooth and it requires no post-processing such as nonlinear filtering or any smoothing processes. Several examples are presented to demonstrate the capability of the harmonics estimation technique and the PDA.

The influence of cochannel, adjacent channel and intermodulation constraints on the capacity of the frequency band in the dynamic channel allocation problem is estimated. Algorithms including a backtracking phase with partial reassignment of currently assigned requirements are proposed. Numerical examples show a strong possibility of a 20% capacity improvement compared to the conventional strategies.