Coaxial-core erbium-doped fiber amplifiers (EDFA's) having a property of self-regulated gain spectrum are developed. The operation of a coaxial-core EDFA is based on the partial separation of the light paths for different wavelength channels in the directionally-coupled waveguides of a coaxial-core geometry. The degree of channel equalization depends on the geometrical and optical parameters of the coaxial-core EDFA and on relative channel power levels. A numerical analysis based on the coupled-mode theory and on the rate equation shows that, under fully optimized conditions, a coaxial-core EDFA provides equalization rates in excess of -0.4 dB per dB of input-power imbalance in the case with two WDM channels. A cascade experiment demonstrates the effect of coaxial-core EDFA's toward channel-power equalization in fiber links with a small number of WDM channels.

In the present paper, a new coding scheme for infrared ASK communication systems--Pulse Sequence Modulation (PSM)--will be proposed, which uses a combination of pulse sequences so that it canattain zero-spectrum points at specific frequencies within the main-lobe region. The interference to the other IR systems was a problem of the conventional infrared ASK communication systems. The proposed PSM coding scheme reduces the interference by decreasing the emission at specific frequencies, and helps multiple IR communication systems co-exist without deteriorating coding efficiency.

Coaxial-core erbium-doped fiber amplifiers (EDFA's) having a property of self-regulated gain spectrum are developed. The operation of a coaxial-core EDFA is based on the partial separation of the light paths for different wavelength channels in the directionally-coupled waveguides of a coaxial-core geometry. The degree of channel equalization depends on the geometrical and optical parameters of the coaxial-core EDFA and on relative channel power levels. A numerical analysis based on the coupled-mode theory and on the rate equation shows that, under fully optimized conditions, a coaxial-core EDFA provides equalization rates in excess of -0.4 dB per dB of input-power imbalance in the case with two WDM channels. A cascade experiment demonstrates the effect of coaxial-core EDFA's toward channel-power equalization in fiber links with a small number of WDM channels.

In the present paper, a new coding scheme for infrared ASK communication systems--Pulse Sequence Modulation (PSM)--will be proposed, which uses a combination of pulse sequences so that it canattain zero-spectrum points at specific frequencies within the main-lobe region. The interference to the other IR systems was a problem of the conventional infrared ASK communication systems. The proposed PSM coding scheme reduces the interference by decreasing the emission at specific frequencies, and helps multiple IR communication systems co-exist without deteriorating coding efficiency.

This paper describes a spatial spectral subtraction method by using the complementary beamforming microphone array to enhance noisy speech signals for speech recognition. The complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this paper, it is shown that the nonlinear subtraction processing with complementary beamforming can result in a kind of the spectral subtraction without the need for speech pause detection. In addition, the optimization algorithm for the directivity pattern is also described. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations under both stationary and nonstationary noise conditions. In comparison with the optimized conventional delay-and-sum (DS) array, it is shown that: (1) the proposed array improves the signal-to-noise ratio (SNR) of degraded speech by about 2 dB and performs more than 20% better in word recognition rates under the conditions that the white Gaussian noise with the input SNR of -5 or -10 dB is used, (2) the proposed array performs more than 5% better in word recognition rates under the nonstationary noise conditions. Also, it is shown that these improvements of the proposed array are same as or superior to those of the conventional spectral subtraction method cascaded with the DS array.

As a means of CDMA network evolution toward future wireless services, a spectral overlay of narrowband CDMA (N-CDMA) and wideband CDMA (W-CDMA) systems is proposed in [8]. In order to justify the overlaying strategy, the reverse link capacity is examined in the same work. Although the capacity of conventional CDMA cellular systems is usually limited by the reverse link, the limit could occur at the forward link depending on the transmission technologies adopted by specific CDMA proposals. Especially, the number of users that can be simultaneously accommodated in the system would be limited by the forward link in future mobile service environments where unequal traffic is offered between two links. In this paper, we first examine the forward link capacity of the spectrally overlaid narrowband and wideband CDMA (N/W CDMA) system. And we compare it with the reverse link capacity to obtain the overall performance. The effects of various parameters on the capacity of N/W CDMA system are numerically evaluated for different mobile environments.

In this paper, we study a new hybrid speech coder which employs a modified version of the harmonic sinusoidal analysis to encode the periodic contents of speech waveform and to split the speech spectrum into two frequency regions of harmonic and random components. A reliable fundamental frequency is estimated for the harmonic region using both speech and its linear predictive (LP) residual spectrum. The peak envelope of speech spectrum is encoded in terms of the coefficients of an all-pole spectrum. A harmonic tracking algorithm appropriately interpolates the sinusoidal parameters to achieve a smooth transition between the parameter update points and to reconstruct an essential level of periodicity in the synthetic voiced speech. The random part of spectrum and unvoiced speech are coded using the conventional CELP algorithm. The individual components are then combined at the decoder to obtain the synthetic speech. The proposed hybrid coder which combines the powerful features of the sinusoidal and CELP coding algorithms yeilds a high quality synthetic speech at 4.05 kbps.

Two classes of nonlinear feedforward logic (NLFFL) pseudonoise (PN) code generators based on the use of AND and majority logic (ML) gates are compared. Cross-correlation and code-division multiple-access (CDMA) properties of properly designed NLFFL sequences are found to be comparable with the properties of well-known linear PN codes. It is determined that code design employing ML gates with an odd number of inputs is easier compared with designing with AND gates. This is especially true when the degree of nonlinearity is large, since the nonbalance problem, e. g. , at the output of an AND gate, can be avoided. ML type sequences are less vulnerable to correlation attack and jamming by the m-sequence of an NLFFL generator

This paper presents a new technique for the synthesis of sets of low-peak sequences exhibiting low peak cross correlation. The sequences also have flat power spectra and are suitable for many applications requiring such sets of uncorrelated pseudo-white-noise sources. This is a new application of the ta-sequence (trigonometric function aliasing sequence), which itself is a very new technique that uses the well-known "Reed-Solomon code" or "One coincident code" to generate these sets of low-peak-factor pseudo-white-noise exhibiting low peak cross correlation. The ta sequence method presented here provides the means for generating various sequences at the lengths required for such applications as system measurement (needing uncorrelated test signals), pseudo-noise synthesis (for spread spectrum communication), and audio signal processing for sound production (for enhancing spatial imagery in stereo signals synthesized from mono sources) and sound reproduction (for controlling unwanted interference effects in multiple-loudspeaker arrays).

We propose an optical spread-time code-division multiple-access (ST-CDMA) with pulse position modulation (PPM) signaling for high-speed communication networks. We obtain a union upper bound on the bit error rate (BER) considering the multi-access interference (MAI), shot noise and thermal noise at the receiver. As a result, we show that the optical ST-CDMA with PPM signaling improves the BER performance at the same received power and bit rate compared to that with OOK signaling. This leads to an increase of the bit rate at the same BER. Moreover, we show that the proposed system can relax the requirement for spectral resolution compared to the optical ST-CDMA with OOK signaling under the received power and bit rate constraints.

We have developed a new type of phase interpolation direct digital synthesizer (DDS) with a symmetrically structured delay generator. The new DDS is similar to a sine output DDS in that it produces lower spurious signals, but it does not require a sine look-up table. The symmetrically structured delay generator reduces the periodic jitter in the most significant bit (MSB) of the DDS accumulator. The symmetrical structure enables the delay generator to produce highly accurate delay timing and eliminates the need to adjust the circuit constants. Experimental results confirm frequency synthesizer operation in which the spurious signal level is reduced to less than that of the accumulator.

The feasibility of a dynamic zone configuration technique has been investigated. To make it easy to implement this technique in wireless communication systems, a simplified method for determining a suitable weight vector by using the least squares (LS) methods was developed. Simulations showed that the developed system is more effective than the present omni-directional zone system. Moreover, combining dynamic zone configuration technique with dynamic channel assignment strategy reduced blocking rate, forced call termination ratio, and required transmission power.

The formulation of the wall admittance of a circular microstrip antenna by the spectral domain method is presented. The circular microstrip antenna is calculated using the cavity model. The electromagnetic fields within the antenna cavity are determined from the impedance boundary condition at the side aperture. The contribution from the region outside the antenna is taken into account by the wall admittance. The wall admittance is defined by the magnetic field produced by the equivalent magnetic current at the aperture. The magnetic field is calculated by the spectral domain method. The wall admittances obtained by this method are compared with the results calculated by Shen. The calculated input impedances of the microstrip antenna agree fairly well with the experimental data for the substrate thickness of up to 0.048λg. The formulation of wall admittance presented here is easily applicable to arbitrarily shaped microstrip antennas.

This paper proposes a method of improving the accuracy of the attenuation constant estimate obtained by using the cross-spectral technique. In the cross-spectral technique, the envelope of the estimated impulse response is deformed due to the use of a time window. As a result, the estimated impulse response decays more rapidly than the real impulse response does, and the attenuation constant obtained by the estimated impulse response becomes larger than the real value. This paper first describes how the attenuation constant changes in the process of impulse response estimation. Next, we propose a method of improving the accuracy of the estimation. The effect of the proposed method is confirmed by computer simulation.

In this paper, the correlation properties are used to develop two efficient encoding schemes for speech line spectrum frequency (LSF) parameters. The first scheme (1D KL), which exploits the intraframe correlation, is based on one-dimensional Karhunen-Loeve (KL) transformation; the second scheme, which requires some coding delays to further utilize the interframe correlation, uses two-dimensional (2D KL) transform in the frequency domain or one-dimensional KL transform co-operating with DPCM in the time domain. Moreover, since the KL transform is globally optimal, which is sensitive to the change of input data statistics, further two adaptive transform coding systems are also investigated in this paper. The performance of all systems for different bit rates is investigated and adequate comparisons are made. It is shown that the gain of using KL transformation to exploit the intraframe and interframe correlation is 3 and 4 bits/speech frame, respectively.

This paper proposes a phase-rotating phase-shift keying (PSK) modulation and shows that its narrow-band version is suitable for Viterbi equalization. The proposed PSK has the following features: 1) a spectrum shaping of the transmit/receive filters does not need to be restricted to the Nyquist criterion; 2) the transmitted data sequence is rotated for every symbol in order to reduce noise-correlation at the receiver. First, this paper discusses a performance degradation of bit error rate of Viterbi equalizers in the presence of the sampling timing offset or under time-dispersive frequency selective fading. Next, computer simulation confirms that π/2-shifted binary PSK with narrow-band spectrum shaping filter, which includes offset QPSK for its special case, solves the above mentioned performance degradation, keeping good spectrum efficiency equal to M-ary PSK.

An Acoustic echo canceller has problems adaptating under noisy or double-talk conditions. The adaptation process requires a precise identification of the temporarily changed room impulse response. To do this, both minimizing the step size parameter of the Least Mean Square (LMS) method to be as small as possible and giving up on updating the adaptive filter coefficients have been considered. This paper describes an adaptive cross-spectral technique that is robust to adaptive filtering under noisy or double-talk conditions and for colored signals such a speech signal. The cross-spectral technique was originally developed to measure the impulse response in a linear system. Here we apply in the adaptive cross-spectral technique to solve the acoustic echo cancelling problem. This cross-spectral technique takes the ensemble average of the cross spectrum between input and error signals and the averaged cross spectrum is divided by the averaged power spectrum of the input signal to update the filter coefficients. We have confirmed that the echo signal is suppressed by about 15 dB even under double-talk conditions. We also explain that this method has a systematic error due to using a short time block for estimating the room impulse response. Then we investigate overlapping every last half block by the following first half block in order to reduce the effect of the systematic error. Finally, we compare our method with the Frequency-domain Block LMS (FBLMS) method because both methods are implemented in the frequency domain using a short time block.

We present a new method for estimation of spectrum transition of nonstationary signals in cases of low signal-to-noise ratio (SNR). Instead of the basic functions employed in the previously proposed time-varying autoregressive (AR) modeling, we introduce a spectrum transition constraint into the cost function described by the partial correlation (PARCORR) coefficients so that the method is applicable to noisy nonstationary signals of which spectrum transition patterns are complex. By applying this method to the analysis of vibration signals on the interventricular septum (IVS) of the heart, noninvasively measured by the novel method developed in our laboratory using ultrasonics, the spectrum transition pattern is clearly obtained during one cardiac cycle for normal subjects and a patient with cardiomyopathy.

Fluctuation characteristics of character centroid intervals in laterally written Japanese sentences are investigated by means of their spatial frequency characteristics. Power spectra of character centroid intervals in their longitudinal and transverse directions are obtained for handwritten and word processor printed sample sentences. It is shown that for fluctuation characteristics in the longitudinal direction, power spectra are inversely proportional to their spatial frequencies for handwritten sentences and proportional to them for word processor printed sentences, and there exists a remarkable difference between handwritten and word processor printed sentences. It is also shown that for fluctuation characteristics in the transverse direction, power spectra are proportional to their spatial frequencies for both handwritten and word processor printed sentences, and there is no remarkable difference between handwritten and word processor printed sentences.

The distributions of a spectral intensity of the breaking arc between silver contacts in DC 45-66 V/2.5-5.0 A circuits have been measured using a high-speed color video. As a result, a cathode brightening spot, which has a high spectral intensity, exists near the cathode surface. The cathode brightening spot expands with the increase of the contact gap, but its length expands until about 18µm. When the contact gap spreads over about 180 µm, a dark positive column appears and grows between the cathode brightening spot and the anode surface. The higher the interrupted current is, the larger the diameter of the cathode brightening spot will be. The maximum diameter of cathode brightening spot is 500 µm under these experiments.