Our study focuses on an evaluation of a novel speech processing strategy for multi-channel cochlear implant speech processors. Stimulation pulse trains for the Nucleus 24CI speech processor were generated in a way different from the speech processing strategies implemented in this processor. The distinctive features of the novel strategy are: 1) electrode stimulation order driven by location of maximum instantaneous frequency amplitude; 2) variable stimulation rates on electrodes; 3) variable number of selected channels within a cycle of signal processing schema. Within-subject designed tests on Japanese initial, medial and final consonants in CV, VCV and CV/N context tokens were carried out with cochlear implant patients using the Cochlear ACETM strategy, and results were compared with those of normal hearing listeners. Results of the initial and medial consonant tests showed significantly better performance with the novel strategy than with the ACE strategy for both the cochlear implant and normal hearing listener groups. Results of the final consonant tests showed a slightly better performance with the ACE strategy for cochlear implant listeners while showing a slightly better performance with the novel strategy for normal hearing listeners.

We describe the first highly nonlinear dispersion-flattened polarization-maintaining photonic crystal fiber designed for nonlinear optics applications in the 1.55 µm region. The nonlinear coefficient of the fiber is 19 (W-1km-1), which is ten times that of dispersion shifted fiber. The chromatic dispersion and dispersion slope of the fiber at 1.55 µm are -0.23 ps/km/nm and 0.01 ps/km/nm2, respectively. We demonstrate the generation of a supercontinuum using the photonic crystal fiber. A symmetrical supercontinuum over 40 nm is obtained by injecting 1562 nm, 2.2 ps, and 40 GHz optical pulses into the 200 m-long photonic crystal fiber.

We propose a simple method for the chromatic dispersion measurement of optical fibers by using bi-directional modulation of a Mach-Zehnder electro-optical modulator embodied in a fiber loop mirror. The detected output of the bi-directionally modulated light, with time difference, creates fading in the RF domain. Dispersion is found by measuring the period of fading at different wavelengths.

An analysis of the phase difference spectrum between two images allows precise image shift detection. Image shifts are directly evaluated from the phase difference spectrum without Fourier inversion. In the calculation, the weight function containing the frequency and the cross spectrum is used and an unlapping procedure is carried out. In an experiment using synthetic and real images of typical image patterns, accuracy as high as 0.01-0.02 pixel was achieved stably and reliably for most of the image patterns.

In this paper, we propose a new method of automatic speech summarization for each utterance, where a set of words that maximizes a summarization score is extracted from automatic speech transcriptions. The summarization score indicates the appropriateness of summarized sentences. This extraction is achieved by using a dynamic programming technique according to a target summarization ratio. This ratio is the number of characters/words in the summarized sentence divided by the number of characters/words in the original sentence. The extracted set of words is then connected to build a summarized sentence. The summarization score consists of a word significance measure, linguistic likelihood, and a confidence measure. This paper also proposes a new method of measuring summarization accuracy based on a word network expressing manual summarization results. The summarization accuracy of each automatic summarization is calculated by comparing it with the most similar word string in the network. Japanese broadcast-news speech, transcribed using a large-vocabulary continuous-speech recognition (LVCSR) system, is summarized and evaluated using our proposed method with 20, 40, 60, 70 and 80% summarization ratios. Experimental results reveal that the proposed method can effectively extract relatively important information by removing redundant or irrelevant information.

A new spectrum management architecture for a flexible software defined radio (SDR) is proposed. In this architecture, the SDR hardware and software are certified separately so as not to destroy the SDR flexibility, but to ensure that any combinations of hardware and software are compliant to the radio regulations even at the system (vertical) handover, global (horizontal) handover, and upgrade (forward) or downgrade (backward) handover. This architecture is based on automatic calibration & certification unit (ACU), built-in GPS receiver, and radio security module (RSM). The ACU is a hardware embedded RF manager that dynamically controls the output power spectrum to be compliant to the local radio regulation parameters. This local radio regulation parameters are securely downloaded to the hardware as an electronic label of the SDR software and stored in the RSM which is a security manager of the hardware. The GPS position check is used, especially during roaming, to keep the compliancy of the terminal to each local radio regulations managed by the geographical region. The principle parties involved in this architecture are telecommunication certification body (TCB), SDR hardware maker (HW maker), SDR software maker (SW maker), and SDR user. The roles and relationships of these four parties in the proposed architecture are clarified in this paper.

SpecC language is designated to handle the design of entire system from specification to implementation and of hardware/software co-design. Concurrency is one of the features of SpecC which expresses the parallel execution of processes. Describing the systems which contain concurrent behaviors would have some data exchanging or transferring among them. Therefore, the synchronization semantics (notify/wait) of events should be incorporated. The actual design, which is usually sophisticated by its characteristic and functionalities, may contain a bunch of event synchronization codes. This will make the design difficult and time-consuming to verify. In this paper, we introduce a technique which helps verifying the synchronization of events in SpecC. The original SpecC code containing synchronization semantics is parsed and translated into a Boolean SpecC code. The difference decision diagrams (DDDs) is used to verify for event synchronization on Boolean SpecC code. The counter examples for tracing back to the original source are given when the verification results turn out to be unsatisfied. Here we also introduce idea on automatically refinement when the results are unsatisfied and preset some preliminary results.

A novel method is presented to compute moments of high-speed VLSI interconnects, which are modeled as coupled RLC trees. Recursive formulae of moments of coupled RC trees are extended to those for coupled RLC trees by considering both self inductances and mutual inductances. Analytical formulae for voltage moments at each node are derived explicitly. The formulae can be efficiently used for estimating delay and crosstalk noise. The inductive crosstalk noise waveform can be accurately and efficiently estimated using the moment computation technique in conjunction with the projection-based order reduction method. Fundamental aspects of the proposed approach are described in details. Experimental results show the increased accuracy of the proposed method over that of the traditional ones.

The discrete nature of data in a functional domain can generally be replaced by the global nature of data in the spectrum domain. In this paper we propose a fast procedure to detect autosymmetric function as an application of the spectrum technique. The autosymmetric function differs from the usual symmetric function and strongly relates with EXOR-based representations. It is known that many practical logical networks are autosymmetric, and this nature allows a useful functional class to realize a compact network with EXOR gates. Our procedure is able to detect autosymmetric functions quickly by using spectral coefficients. In experiments, our technique can detect the autosymmetry of most networks with a small number of checks of the spectrum.

We analyze the timing design methodology for testing chips using a multiple-clock domain scheme. We especially focus on the layout design of the design-for-test (DFT) circuits and the clock network. First, we demonstrate the built-in-self-testing (BIST) scheme for multiple-clock domains. Then, we discuss the layout method that achieves a low clock-skew between different clock domains with a small modification of the original user logic layout. Finally, we evaluate the fault coverage of our large ASIC chips designed using our new methodology. The short design period and high fault coverage of our methodology are confirmed using actual industrial designs. We introduce a viable approach for industrial designs because designers don't have to pay much attention to DFT. Our approach also provides designers with an easy method for LSI debugging and diagnostics.

This paper proposes a packet loss recovery method using packets arrived behind the playout time for CELP (Code Excited Liner Prediction) decoding. The proposed method recovers synchronization of the filter states between encoding and decoding in the period following packet loss. The recovery is performed by replacing the degraded filter states with the ones calculated from the late arrival packet in decoding. When the proposed method is applied to the AMR (Adaptive Multi-Rate) speech decoder, it improves the segmental SNR (Signal-to-Noise Ratio) by 0.2 to 1.8 dB at packet loss rates of 2 to 10 % in case that all the packet losses occur due to their late arrival. PESQ (Perceptual Evaluation of Speech Quality) results also show that the proposed method slightly improves the speech quality. The subjective test results show that five-grade mean opinion scores are improved by 0.35 and 0.28 at a packet loss rate of 5 % at speech coding bitrates of 7.95 and 12.2 kbit/s, respectively.

Analysis of concurrent systems, such as computer/communication networks and manufacturing systems, usually employs formal discrete event models. The analysis then includes model validation, property verification, and performance evaluation of such models. The DEVS (Discrete Event Systems Specification) formalism is a well-known formal modeling framework which supports specification of discrete event models in a hierarchical, modular manner. While validation and verification using formal models may not resort to discrete event simulation, accurate performance evaluation must employ discrete event simulation of formal models. Since formal models, such as DEVS models, explicitly represent communication semantics between component models, their simulation cost is much higher than using simulation languages with informal models. This paper proposes a method for simulation speedup in performance evaluation of concurrent systems using DEVS models. The method is viewed as a compiled simulation technique which eliminates run-time interpretation of communication paths between component models. The elimination has been done by a behavior-preserved transformation method, called model composition, which is based on the closed under coupling property in DEVS theory. Experimental results show that the simulation speed of transformed DEVS models is about 14 times faster than original ones.

We have developed an OC-48c (2.4 Gbps) PCI-compliant network interface card and drivers with the aim of evaluating the effectiveness of our proposed link layer protocol MAPOS. In this paper, we study the effectiveness of MAPOS particularly from the viewpoint of the influence of packet sizes up to the 64-kbyte jumbo MTU size and the effectiveness of our new implementation of the non-interrupt-driven sending process and interrupt batching receiving process deployed to improve the throughput in short-packet transmissions. Our main findings are as follows; Enlarging the packet size up to 64-kbyte MTU improves the performance in transmission. OC-48c wire speed is achieved with packet sizes larger than 16 kbytes. Implementation of the non-interrupt-driven sending process and the interrupt batching receiving process improves the performance of short-packet transmission. In particular, the transmission throughput is improved by 50% when 64-byte short packets are used. The maximum loss-free receive rate is also raised by 50% when 4-kbyte packets arrive. With a high-performance CPU, the data-transfer speed of the DMA controller for jumbo packets cannot keep up with the packet-queueing speed of the CPU. Our proposed procedure for adaptive algorithm switching method can resolve this problem. The maximum TCP throughput observed in our measurement using the latest PCs and MAPOS OC-48c PCI card was 2342.5 Mbps. This throughput represents the highest performance in a legacy-PCI-based system according to the results database of the benchmarking software.

The adaptive cross-spectral (ACS) technique recently introduced by Okuno et al. provides an attractive solution to acoustic echo cancellation (AEC) as it does not require double-talk (DT) detection. In this paper, we first introduce a generalized ACS (GACS) technique where a step-size parameter is used to control the magnitude of the incremental correction applied to the coefficient vector of the adaptive filter. Based on the study of the effects of the step-size on the GACS convergence behaviour, a new variable step-size ACS (VSS-ACS) algorithm is proposed, where the value of the step-size is commanded dynamically by a special finite state machine. Furthermore, the proposed algorithm has a new adaptation scheme to improve the initial convergence rate when the network connection is created. Experimental results show that the new VSS-ACS algorithm outperforms the original ACS in terms of a higher acoustic echo attenuation during DT periods and faster convergence rate.

In speech enhancement with adaptive microphone array, the voice activity detection (VAD) is indispensable for the adaptation control. Even though many VAD methods have been proposed as a pre-processor for speech recognition and compression, they can hardly discriminate nonstationary interferences which frequently exist in real environment. In this research, we propose a novel VAD method with array signal processing in the wavelet domain. In that domain we can integrate the temporal, spectral and spatial information to achieve robust voice activity discriminability for a nonstationary interference arriving from close direction of speech. The signals acquired by microphone array are at first decomposed into appropriate subbands using wavelet packet to extract its temporal and spectral features. Then directionality check and direction estimation on each subbands are executed to do VAD with respect to the spatial information. Computer simulation results for sound data demonstrate that the proposed method keeps its discriminability even for the interference arriving from close direction of speech.

A two stage non-scan design for testability method is proposed. The first stage selects test points based on an earlier testability measure conflict. A new design for testability algorithm is proposed to select test points by a fault-oriented testability measure conflict+ in the second stage. Test points are selected in the second stage based on the hard faults after the initial ATPG run of the design for testability circuit in the preliminary stage. The new testability measure conflict+ based on conflict analysis of hard-faults in the process of test generation is introduced, which emulates most general features of sequential ATPG. The new testability measure reduces testability of a fault to the minimum D or controllability of the primary outputs, and therefore, does not need observability measure any more. Effective approximate schemes are adopted to get reasonable estimation of the testability measure. A couple of effective techniques are also adopted to accelerate the process of the proposed design for testability algorithm. Experimental results show that the proposed method gets even better results than two of the recent non-scan design for testability methods nscan and lcdft.

A novel hybrid numerical method, which is based on the extended spectral domain approach combined with the mode-matching method, is applied to evaluate the scattering parameter of waveguide discontinuities. The formulation procedure utilizes the biorthogonal relation in the transformation, and the Green's functions in the spectral domain are obtained easily even in the inhomogeneous lossy regions. The present method does not include the approximate perturbational scheme, and it can evaluate accurately and stably the scattering parameters of either for the thin or thick obstacles made of the wide variety of materials, the lossless dielectrics to highly conductive media, in short computation time. The physical phenomena of transmission through the lossy obstacles are investigated by numerical computations. The results are compared with FEM where FEM computations are feasible, although the FEM computations cannot cover the whole performances of the present method. The good agreement is observed in the corresponding range. The matrix size in this method is smaller than that of other methods. Therefore, the present method is numerically efficient and it would be able to apply for the integrated evaluation of a successive discontinuity. The resonant characteristics of rectangular waveguide cavity are analyzed accurately taking the conductor losses into consideration.

A high-speed 3-D camera has a future possibility of wide variety of application fields such as quick inspection of industrial components, observation of motion/destruction of a target object, and fast collision prevention. In this paper, a row-parallel position detector for a high-speed 3-D camera based on a light-section method is presented. In our row-parallel search method, the positions of activated pixels are quickly detected by a row-parallel search circuit in pixel and a row-parallel address acquisition of O(log N) cycles in N-pixel horizontal resolution. The architecture keeps high-speed position detection in high pixel resolution. We have designed and fabricated the prototype position sensor with a 12816 pixel array in 0.35 µm CMOS process. The measurement results show it achieves quick activated-position acquisition of 450 ns for "beyond-real-time" 3-D imaging and visual feedback. The high-speed position detection of the scanning sheet beam is demonstrated.

In an optical time domain reflectometer type strain measurement system, we theoretically derive the shape of the Brillouin gain spectrum produced in an optical fiber under a parabolic strain distribution which is formed in a uniformly loaded beam. Based on the derived result, we investigate the effects of the parabolic strain distribution parameters and the measurement conditions such as the launched pulse width and the measurement position on the beam on the deformation of the Brillouin backscattered-light power spectrum shape. In addition, we investigate the strain measurement error resulting from the deformation of the power spectrum shape by analyzing the peak-power frequency at which the power spectrum is maximized.

In this paper, we describe the operation of circuits capable of more than 40-Gbit/s that we have developed using InP HEMT technology. For example, we succeeded in obtaining 43-Gbit/s operation for a full-rate 4:1Multiplier (MUX), 50-Gbit/s operation for a Demultiplexer (DEMUX), 50-Gbit/s operation for a D-type flip-flop (D-FF), and a preamplifier with a bandwidth of 40 GHz. In addition, the achievement of 90-Gbit/s operation for a 2:1MUX and a distributed amplifier with over 110-GHz bandwidth indicates that InP HEMT technology is promising for system operations of over 100 Gbit/s. To achieve these results, we also developed several design techniques to improve frequency response above 80 GHz including a symmetric and separated layout of differential elements in the basic SCFL gate and inverted microstrip.