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.

This letter presents a method which attempts to minimize the number of spill codes to resolve usage conflicts of distributed registers in application specific DSPs. It searches for a set of ordering restrictions among operations which sequentialize the lifetimes of the values residing in the same register as much as possible. Experimental results show that the proposed analysis method reduces the number of register spills into 28%.

We demonstrate, through numerical simulations, the possibility of trans-oceanic single channel transmission at 160 Gbit/s with no active control. This was achieved using short period dispersion management, which supports short pulse propagation at practical map strengths. We demonstrate that through careful selection and optimisation of the system parameters the performance of this system can be extended. We also define the tolerable limits of the system to the residual dispersion slope and polarisation mode dispersion.

Based on filtering ground clutter power directly in the frequency domain, a new non-coefficient Adaptive MTI (AMTI) scheme is presented in this letter. The results of simulation example show that this scheme has smaller signal-to-noise ratio loss than the classical AMTI based on spectral estimation, as well as high improvement factor.

The demonstrated capacity of 40-Gb/s WDM systems is now exceeding 3-Tb/s. We will discuss some of the enabling technologies for these high capacities and some of the technologies that may push the aggregate capacity of WDM systems towards 10-Tb/s.

We investigated the characteristics of optical duobinary signals in achieving high fiber input power transmission focusing on the idea of optimum residual dispersion equalization. We confirm through calculations and experiments that setting the total link dispersion at a non-zero value allows high fiber launched power (+18 dBm) and large dispersion tolerance (350 ps/nm) at 10 Gbit/s. We demonstrate repeaterless 250-km single mode fiber (SMF) transmission with a 10-Gbit/s optical duobinary signal. We also demonstrate high-speed complete optical duobinary coding and transmit synchronous digital hierarchy (SDH) frames over optical duobinary signals for the first time.

We investigated the characteristics of optical duobinary signals in achieving high fiber input power transmission focusing on the idea of optimum residual dispersion equalization. We confirm through calculations and experiments that setting the total link dispersion at a non-zero value allows high fiber launched power (+18 dBm) and large dispersion tolerance (350 ps/nm) at 10 Gbit/s. We demonstrate repeaterless 250-km single mode fiber (SMF) transmission with a 10-Gbit/s optical duobinary signal. We also demonstrate high-speed complete optical duobinary coding and transmit synchronous digital hierarchy (SDH) frames over optical duobinary signals for the first time.

We demonstrate, through numerical simulations, the possibility of trans-oceanic single channel transmission at 160 Gbit/s with no active control. This was achieved using short period dispersion management, which supports short pulse propagation at practical map strengths. We demonstrate that through careful selection and optimisation of the system parameters the performance of this system can be extended. We also define the tolerable limits of the system to the residual dispersion slope and polarisation mode dispersion.

Using the chirped grating with temperature control, we demonstrated the adaptive dispersion compensation at 40 Gbit/s RZ transmission. The simple monitoring of the 40 GHz frequency component enables us to automatic control of the adaptive dispersion compensator.

This paper conducts performance evaluation and performs simulation for a code division multiple access (CDMA) system when channel bands of multiple neighboring CDMA/DSSS are overlapped in time domain. It is assumed that all systems adopt direct-sequence spread-spectrum (DSSS) technique and are BPSK modulated by the different carrier frequencies. Automatic power control (APC) is also applied in the interfered system such that the receiver gets the same power from all users. Without loss generality, an additive white Gaussian noise (AWGN) channel is also assumed during analysis. In this paper, the analytic solution of the signal to noise ratio (SNR) is first derived in which both CDMA systems are modulated by different carrier frequencies. We have the results by simulation with Δ f = 0 and Δ f = 1 MHz, respectively. This analysis is good for general cases; and the results show an excellent computational performance. In particular, the result is very close to Pursley's result, when the systems have the same code length with no carrier difference.

The demonstrated capacity of 40-Gb/s WDM systems is now exceeding 3-Tb/s. We will discuss some of the enabling technologies for these high capacities and some of the technologies that may push the aggregate capacity of WDM systems towards 10-Tb/s.

Using the chirped grating with temperature control, we demonstrated the adaptive dispersion compensation at 40 Gbit/s RZ transmission. The simple monitoring of the 40 GHz frequency component enables us to automatic control of the adaptive dispersion compensator.

To develop a smoothing method for speckle reduction is a significant problem, because of the complex ultrasonic characteristics and the obscurity of the tissue image. This paper presents a new method for speckle reduction from medical ultrasonic image by using fuzzy morphological speckle reduction algorithm (FMSR) that preserves resolvable details while removing speckle in order to cope with the ambiguous and obscure ultrasonic images. FMSR creates a cleaned image by recombining the processed residual images with a smoothed version of an original image. Performance of the proposed method has been tested on the phantom and tissue images. The results show that the method effectively reduces the speckle while preserving the resolvable details.

We propose a dynamic assignment method of radio channels using their qualities and data packet sizes. In this method, a high quality channel is assigned for long packets and a low quality channel is assigned for short packets. Assuming a packet transfer method based on PHS (Personal Handy System) procedure, the transfer speed and delay characteristics are investigated by computer simulation and compared with conventional method (i.e. random assignment). Both the transfer speed characteristics and delay performance are improved under light traffic load.

A new speech enhancement technique is proposed assuming that a speech signal is represented in terms of a linear probabilistic process and that a noise signal is represented in terms of a stationary random process. Since the target signal, i.e., speech, cannot be represented by a stationary random process, a Wiener filter does not yield an optimum solution to this problem regarding the minimum mean variance. Instead, a Kalman filter may provide a suitable solution in this case. In the Kalman filter, a signal is represented as a sequence of varying state vectors, and the transition is dominated by transition matrices. Our proposal is to construct the state vectors as well as the transition matrices based on time-frequency pattern of signals calculated by a wavelet transformation (WT). Computer simulations verify that the proposed technique has a high potential to suppress noise signals.

The hardware implementation of a proposed high dimensional discrete torus knot code was successfully realized on an ASIC chip. The code has been worked on for more than a decade since then at Aichi Prefectural University and Nagoya Institutes of Technology, both in Nagoya, Japan. The hardware operation showed the ability to correct the errors about five to ten times the burst length, compared to the conventional codes, as expected from the code configuration and theory. The result in random error correction was also excellent, especially at a severely degraded error rate range of one hundredth to one tenth, and also for high grade characteristic exceeding 10-6. The operation was quite stable at the worst bit error rate and realized a high speed up to 50 Mbps, since the coder-decoder configuration consisted merely of an assemblage of parity check code and hardware circuitry with no critical loop path. The hardware architecture has a unique configuration and is suitable for large scale ASIC design. The developed code can be utilized for wider applications such as mobile computing and qualified digital communications, since the code will be expected to work well in both degraded and high grade channel situations.

We have proposed and implemented a spread spectrum (SS) wireless switch using 2.4 GHz front-end AlN/Al2O3 surface acoustic wave (SAW) matched filter (MF). Since the SAW MF has radio frequency (RF) front-end operation, RF components are not needed in the received circuit. High impedance in the peripheral circuit using passive devices has been employed for low current consumption. The SS wireless switches have been designed with the power consumption of less than 100 µW by using the SAW MF. It is confirmed that implemented SS wireless switch has a long battery life of 10 years and communication range of 30 m.

This paper describes experimental results on whole word HMM-based speech recognition of connected digits in Japanese with special focus on the training data size and the "sheep and goats" problem. The training data comprises 757000 digits uttered by 2000 speakers, while the testing data comprises 399000 digits uttered by 1700 speakers. The best word error rate for unknown length strings was 1.64% obtained using context dependent HMMs. The word error rate was measured for various subsets of the training data reduced both in the number of speakers (s) and the number of utterances per speakers (u). As a result, an empirical formula of s[{min(0.62s0.75, u)}0.74 + {max(0, u-0.62s0.75)}0.27] = D(Ew) was developed, where Ew and D(Ew) designate word error rate and effective data size, respectively. Analyses were conducted on several aspects of the low performance speakers accounting for the major part of recognition errors. Attempts were also made to improve their recognition performance. It was found that 33% of the low performance speakers are improved to the normal level by speaker clustering centered around each low performance speaker.

A chemical shift MR method which utilizes a oscillating gradient field is presented in this paper. Frequency modulation resulting from oscillating a gradient field spreads the spectrum that contains both chemical shift and spatial information, over a wide frequency range by using a large modulation factor in FM. The chemical shift spectrum resides within every frequency band segmented by the modulation frequency ωm. The spectral elements gathered from all such frequency segments for a chemical shift frequency contain the spatial image of that particular chemical shift frequency, despite the distortion introduced by a series of the Bessel functions acting as a point spread function. A sum of several Bessel functions of the first kind Jn(. ) is used to approximate the deconvolution process, since the sum staggered with respect to n has a desirable peaking property useful in deconvolution. This leads to devise a new image reconstruction algorithm based on the simple moving average over the spatial coordinate for which the oscillating gradient is applied. Furthermore, the number of echo measurements necessary for an image size of N N is reduced from N2 of the spin echo chemical shift imaging down to N by this method. Simulation results supporting the validity of this method are also presented in this paper.

This paper deals with the problem of autoregressive (AR) spectral estimation from a finite set of noisy observations without a priori knowledge of additive noise power. A joint technique is proposed based on the high-order and true-order AR model fitting to the observed noisy process. The first approach utilizes the uncompensated lattice filter algorithm to estimate the parameters of the over-fitted AR model and is one-pass. The latter uses the noise compensated low-order Yule-Walker (LOYW) equations to estimate the true-order AR model parameters and is iterative. The desired AR parameters, equivalently the roots, are extracted from the over-fitted model roots using a root matching technique that utilizes the results obtained from the second approach. This method is highly accurate and is particularly suitable for cases where the system of unknown equations are strongly nonlinear at low SNR and uniqueness of solution from the LOYW equations cannot be guaranteed. In addition, fuzzy logic is adopted for calculating the step size adaptively with the cost function to reduce the computational time of the iterative total search technique. Several numerical examples are presented to evaluate the performance of the proposed scheme in this paper.