We proposed a simple technique for measuring the effective zero-dispersion wavelength. In this study, we measured the effective zero-dispersion wavelength of a 25-km-long dispersion-shifted fiber (DSF) using the four-wave mixing (FWM) of a spectrum-sliced fiber amplifier light source, and then compared our results with other conventional techniques to confirm the validity of our method.

InP-based high electron mobility transistors (HEMTs) with gate lengths reduced to 30 nm were fabricated and characterized, and the effect of the gate recess on the high-frequency characteristics was studied. The cutoff frequency, which is regarded as a function of the gate length and the average carrier velocity in a first-order approximation, depends on the size of the gate recess when the gate length becomes short. The size of the gate recess is optimized by taking the feed-back capacitance and the parasitic resistance into account. For HEMTs having the gate recess with an InP surface, an appropriate widening of the gate recess gives a record cutoff frequency of 368 GHz for the 30-nm-gate HEMTs with a lattice-matched channel.

This paper proposes a precise method of realizing simultaneous measurement of microscopic defects and the macroscopic three-dimensional shapes of planar objects having specular reflection surfaces. The direction vector field of surface tilt is evaluated directly by the introduction of a moving slit-light technique based on computer graphic animation. A reflected image created by the moving slit-light is captured by a video camera, and the image sequence of the slit-light deformation is analyzed. The obtained direction vector field of the surface tilt recovers the surface shape by means of integration. Two sample objects, a concave mirror and a plane plastic injection molding, are tested to measure the performance of the proposed method. Surface anomalies such as surface dent and warpage are detected quantitatively at a high resolution (about 0.2 [µm]) and a high accuracy (about 95%) in a wide area (about 15 [cm]) of the test object.

This paper proposes a bandwidth division type parallel combinatory (PC) spread spectrum (SS) modulation scheme. In the proposed system, a given system bandwidth for the conventional single-carrier PC-SS system is divided into H subbands, and H PC-SS signals are transmitted in parallel. We evaluate the frame error rate (FER) of the proposed system under the asynchronous CDMA environment. We show that the proposed scheme provides a smaller FER than the single-carrier PC-SS system for a given information bit rate. We also show that the proposed scheme attains a higher information bit rate than the single-carrier PC-SS system for a given FER.

This paper describes the recent progress in the device performance of the uni-traveling-carrier photodiode (UTC-PD). The UTC-PD utilizes only electrons as the active carriers and this unique feature is the key to achieving excellent high-speed and high-output characteristics simultaneously. The achieved performance includes a record 3-dB bandwidth (f3dB) of 310 GHz, a high output current over 180 mA with an f3dB of 65 GHz, a high linearity of up to 80 mA, and a zero-bias operation with an f3dB of 230 GHz and an output peak current of 6.8 mA.

The millimeter wave filter using two whispering-gallery mode dielectric disk resonators is presented in this paper. The coupling coefficients of dual disk resonators and the external Q values of the single resonator excited by a dielectric waveguide are investigated theoretically and experimentally. A 2-stage bandpass filter which is designed at the center frequency of 69.85 GHz with a bandwidth of 500 MHz shows a low-loss property of 1.8 dB insertion loss.

A high quality speech synthesis technique based on the wavelet subband analysis of speech signals was newly devised for enhancing the naturalness of synthesized voiced consonant speech. The technique reproduces a speech characteristic of voiced consonant speech that shows unvoiced feature remarkably in the high frequency subbands. For mixing appropriately the unvoiced feature into voiced speech, a noise inclusion procedure that employed the discrete wavelet transform was proposed. This paper also describes a developed speech synthesizer that employs several random fractal techniques. These techniques were employed for enhancing especially the naturalness of synthesized purely voiced speech. Three types of fluctuations, (1) pitch period fluctuation, (2) amplitude fluctuation, and (3) waveform fluctuation were treated in the speech synthesizer. In addition, instead of a normal impulse train, a triangular pulse was used as a simple model for the glottal excitation pulse. For the compensation for the degraded frequency characteristic of the triangular pulse that overdecreases than the spectral -6 dB/oct characteristic required for the glottal excitation pulse, the random fractal interpolation technique was applied. In order to evaluate the developed speech synthesis system, psychoacoustic experiments were carried out. The experiments especially focused on how the mixed excitation scheme effectively contributed to enhancing the naturalness of voiced consonant speech. In spite that the proposed techniques were just a little modification for enhancing the conventional LPC (linear predictive coding) speech synthesizer, the subjective evaluation suggested that the system could effectively gain the naturalness of the synthesized speech that tended to degrade in the conventional LPC speech synthesis scheme.

This paper proposes a hardware configuration using only single pole dual throw (SPDT) switches to realize the previously proposed automatic calibration method using transmitting signals (ACT) for the adaptive array in TDD communication systems. The proposed configuration obtains the same calibration values as the conventional ACT does while reducing the number of switch branches. The transmission pattern using the proposed calibration method is also presented based on an experimental adaptive array testbed in an actual microcell environment. The experimental results show that the ideal radiation pattern formation is achieved by employing the proposed calibration method in an environment with a moving terminal station and where arriving co-channel interference exists.

In this paper, a high-speed PLA based on dynamic array logic circuits with latch sense amplifiers is presented. The present circuit consists of logic cell arrays, dual-rail bit-lines, latch sense amplifiers, and control blocks. By using a charge sharing scheme and latch sense amplifiers, voltage swings of the bit-lines are reduced compared to the conventional circuits, thus a high-speed and low-power operation is achieved. The present array logic configuration can realize any logic function expressed in the sum-of-products form by using PLA structure. As an application of the proposed PLA, a 32-bit binary comparator is designed and implemented in a 0.6-µm double-poly triple-metal CMOS process. Results of HSPICE simulation show a better performance compared to the conventional circuits. Functional testing using electron beam probing shows that the present circuit operates correctly.

This work presents a novel description of texture mapping polygons in a geometric view about scanlines and a simplified mapping function to improve the performance. The conventional perspective-correct mapping requires costly division operations. In this work, two concepts in perspective geometry, cross-ratio and vanishing point, are exploited to simplify the mapping function. We substitute the point at infinity on scanline into the cross-ratio equation, then obtain a simple description of perspective mapping in polygons. Our mapping function allows the spatial mapping of a pixel from a scanline on a screen plane to a texture plane taking only one division, one multiplication and three additions. The proposed algorithm speeds up the mapping process without losing any correctness. Experimental results indicate that the performance of proposed method is superior to other correct mapping methods.

Sector antennas provide many advantages such as when combined with a narrow beam antenna, they become particularly effective in achieving high-speed wireless communication systems and they aid in simplifying the structure. These antennas have a drawback in that as the number of sectors increases, the antenna size rapidly increases. Therefore, downsizing the sector antenna has become a major research topic. A promising candidate is utilizing a phased-array type antenna; however, this antenna requires a phase-shifter circuit for beam scanning and generally the feeding circuit for this type of antenna is very complicated. To address these issues, we propose a self-selecting feeding circuit that is controlled by the same control circuit and is operated similarly to the conventional single port n-th throw (SPNT) switch. We fabricated a small cylindrical 12-sector antenna at 19 GHz employing the proposed feeding circuit for verification purposes. Furthermore, this paper clarifies the design method of this feeding circuit where the antenna diameter is 71 mm, and the results clearly show that the gain is more than 12 dBi.

In this paper, we propose new approaches to speech enhancement based on soft decision. In order to enhance the statistical reliability in estimating speech activity, we introduce the concept of a global speech absence probability (GSAP). First, we compute the conventional speech absence probability (SAP) and then modify it according to the newly proposed GSAP. The modification is made in such a way that the SAP has the same value of GSAP in the case of speech absence while it is maintained to its original value when the speech is present. Moreover, for improving the performance of the SAP's at voice tails (transition periods from speech to silence), we revise the SAP's using a hang-over scheme based on the hidden Markov model (HMM). In addition, we suggest a robust noise update algorithm in which the noise power is estimated not only in the periods of speech absence but also during speech activity based on soft decision. Also, for improving the SAP determination and noise update routines, we present a new signal to noise ratio (SNR) concept which is called the predicted SNR in this paper. Moreover, we demonstrate that the discrete cosine transform (DCT) enhances the accuracy of the SAP estimation. A number of tests show that the proposed method which is called the speech enhancement based on soft decision (SESD) algorithm yields better performance than the conventional approaches.

In this paper, a novel adaptive digital watermarking approach based upon human visual system model and fuzzy clustering technique is proposed. The human visual system model is utilized to guarantee that the watermarked image is imperceptible. The fuzzy clustering approach has been employed to obtain the different strength of watermark by the local characters of image. In our experiments, this scheme allows us to provide a more robust and transparent watermark.

EMI coupling paths in an electronic controller are investigated experimentally. Common-mode current measurements on the attached cable are used for diagnosing changes made to the EMI coupling path. Experiments that include shielding various portions of the PCB, and re-routing high-speed traces are conducted to characterize the coupling path. A means of identifying and characterizing EMI coupling paths in functioning hardware, and relating them to design features, is demonstrated.

Most of the previous controllers proposed for output regulation problems on uncertain nonlinear systems tried to keep the state variables to the nominal equilibrium points. In this letter, however, the dynamic state feedback controller makes the state variables follow the perturbed equilibrium points computed from an equilibrium-estimator.

This paper elucidates the most appropriate hybrid automatic-repeat-request (ARQ) scheme, i.e., which can achieve the highest throughput, for high-speed packet transmission in the W-CDMA forward link by comparing the throughput performance of three types of hybrid ARQ schemes: type-I hybrid ARQ with packet combining (PC), type-II hybrid ARQ, and basic type-I hybrid ARQ as a reference. Moreover, from the viewpoint of maximum throughput, the respective optimum roles of ARQ and channel coding in hybrid ARQ are also clarified, such as the optimum coding rate and the packet length related to the interleaving effect. The simulation results reveal that the type-II scheme exhibits the best throughput performance, and the required received signal energy per chip-to-background noise spectral density ratio (Ec/N0) at the throughput efficiency of 0.2/0.4/0.6 is improved by 0.7/0.3/0.1 dB and 3.9/1.8/0.5 dB, respectively, compared to the type-I scheme with and without PC in a 2-path Rayleigh fading channel with the average equal power at the maximum Doppler frequency of 5 Hz and the packet length of 4 slots (= 0.667 4 = 2.667 msec). However, the improvement of the type-II scheme compared to the type-I scheme with PC is small or the achievable throughput is almost identical in the high-received Ec/N0 region. On the other hand, the type-I scheme with PC is much less complex and thus preferable, while maintaining almost the same throughput performance or allowing very minor degradation compared to that with type-II. The results also elucidate that, while the optimum coding rate depends on the required throughput in the basic type-I and type-I with PC schemes, it is around between 3/4 and 8/9 in type-II, resulting in a higher throughput efficiency. In addition, for high-speed packet transmission employing a hybrid ARQ scheme, a shorter retransmission unit size is preferable such as 1 slot, and the fast transmit power control is effective only under conditions such as a low maximum Doppler frequency and a high transmit Ec/N0 region.

This paper presents a new approach to modeling speech spectra and pitch for text-independent speaker identification using Gaussian mixture models based on multi-space probability distribution (MSD-GMM). MSD-GMM allows us to model continuous pitch values of voiced frames and discrete symbols for unvoiced frames in a unified framework. Spectral and pitch features are jointly modeled by a two-stream MSD-GMM. We derive maximum likelihood (ML) estimation formulae and minimum classification error (MCE) training procedure for MSD-GMM parameters. The MSD-GMM speaker models are evaluated for text-independent speaker identification tasks. The experimental results show that the MSD-GMM can efficiently model spectral and pitch features of each speaker and outperforms conventional speaker models. The results also demonstrate the utility of the MCE training of the MSD-GMM parameters and the robustness for the inter-session variability.

A new approach for extracting significant characteristic within speech signal for distinct speaker is presented. Based on the multiresolution property of wavelet transform, quadrature mirror filters (QMFs) derived by Daubechies is used to decompose the input signal into varied frequency channels. Owning to the uncorrelation property of each resolution derived from QMFs, Linear Predict Coding Cepstrum (LPCC) of lower frequency region and entropy information of higher frequency region for each decomposition process are calculated as the speech feature vectors. In addition, a hard thresholding technique for lower resolution in each decomposition process is also used to remove the effect of noise interference. The experimental result shows that by using this mechanism, not only effectively reduce the effect of noise inference but improve the recognition rate. The proposed feature extraction algorithm is evaluated on MAT telephone speech database for Text-Independent speaker identification using vector quantization (VQ). Some popular existing methods are also evaluated for comparison in this paper. Experimental results show that the performance of the proposed method is more effective and robust than that of the other existing methods. For 80 speakers and 2 seconds utterance, the identification rate is 98.52%. In addition, the performance of our method is very satisfactory even at low SNR.

A small data-line-swing read/write scheme is described for half-Vcc plate nonvolatile DRAMs with ferroelectric capacitors designed to achieve high reliability for read/write operations. In this scheme, the normal read/write operation holds the data as a charge with a small data-line-swing, and the store operation provides sufficient polarization with a full data-line-swing. This scheme enables high read/write endurance, because the small data-line-swing reduces the fatigue of the ferroelectric capacitor. Two circuit technologies are used in this scheme to increase the operating margin. The first is a plate voltage control technique that solves the polarization retention problem of half-Vcc plate nonvolatile DRAM technologies. The second is a doubled data-line-capacitance recall technique that connects two data lines to a cell and enlarges the readout signal compared to normal operation, when only one data line is connected to a cell. These techniques and circuits improve the write-cycle endurance by almost three orders of magnitude, while reducing the array power consumption during read/write operations to one-third that of conventional nonvolatile DRAMs.

In this paper, we first propose a new speech enhancement preprocessing algorithm by combining power subtraction method and maximal ratio combining technique, then apply it to both energy-based and statistical model-based VAD algorithm to improve the performance even in low SNR conditions. We also perform extensive computer simulations to demonstrate the performance improvement of the proposed VAD algorithm employing the proposed speech enhancement preprocessing algorithm under various background noise environments.