A low-loss serial power combiner using suspended stripline is described. It consists of novel broadside-coupled directional couplers which have shunt capacitances at the edges of the coupled sections. These additional shunt capacitances compensate for poor directivities of the couplers because of inhomogeneous dielectric in suspended stripline structure. The fabricated three-way power combiner has achieved good performance with insertion loss less than 0.23 dB over a bandwidth of 10% in 2 GHz band.

Some plants have air purification ability. This purification ability of plants is considered a promising method for indoor air purification because of the low cost and high purification performance. Therefore, several studies have been carried out to investigate the relationship between the air purification ability of plants and environmental conditions. Nevertheless, the purification mechanism and process have not been clarified yet. In this paper, we investigated the air purification process in plants by bioelectrical potential analysis using linear and nonlinear analysis methods. First, we showed that two types of plants have a high air purification ability; Schefflera and Boston fern. Next, we measured AC bioelectrical potential during the purifying process of plants for pollutant gas. Then, we evaluated the power spectra of time series data of the bioelectrical potential. We found that the power spectra shifted to a lower level after gas injection over all frequencies. Thus, the higher power spectrum came from possible higher physiological activities of the plant. Finally, we introduced a nonlinear analysis method from the dynamical system theory. We transformed the time series data of the potential to a higher dimensional state space using a delay coordinate, which is often used in the field of nonlinear time series analysis. The results show that the orbits in the reconstructed state space have a large variation in gas injection. These experimental results suggest that the measurement of bioelectrical potential could become a useful method for evaluating the air purification ability of plants.

Due to the cost of multicast state management, multicast address allocation, inter-domain multicast routing of traditional IP multicast scheme, ASM leads to a search for other multicast schemes. This paper presents a new solution to the problems mentioned above based on IPv6. The proposed scheme provides an enhanced scheme supporting the strengths of SSM in basic Xcast. This is achieved by adding MLDv2 operations at recipient's side and a new control plane into existing Xcast. The proposed scheme does not only provide the transparency of traditional multicast schemes to sources and recipients, but it also enhances the routing efficiency in networks. Intermediate routers do not have to maintain multicast state, so that it results in a more efficient and scalable mechanism to deliver native multicast datagrams. Also, the seamless integration in Mobile IPv6 can support multicast efficiently for mobile nodes in IPv6 networks by avoiding tunnel avalanches and tunnel convergence. We've attempted to prove this alternative architecture by both simulation and implementation, respectively. Our approach cannot fundamentally perform for many large groups distributed widely as effectively as traditional multicast schemes. However, we believe that the resulting scheme is simple, efficient, robust, transparent, and to the extent possible, scalable in case that recipients are clustered in subnets.

We have determined the thickness and optical constants (refractive index and extinction coefficient) of each layer in the multi-layer organic light emitting diode (OLED) devices based on phosphorescent platinum octaethyl porphine (PtOEP) using a phase modulated spectroscopic ellipsometer. The thickness of each layer estimated from the ellipsometric measurement is different from the thickness measured with quartz oscillator during the evaporation of organic materials. The deviation of total multi-layer thickness is about 5%, while the deviation in each of N, N'-bis(1-naphtyl)-N, N'-diphenyl-1,1'-biphenyl-4,4'-diamine (α-NPD) and aluminum tris 8-hydroxyquinoline (Alq3) layers is about 20-25%. Additionally the spectra of refractive index and extinction coefficient of Alq3 and α-NPD layers are different from those that are measured using the single layer films. These results are understood by penetration of organic material from the neighboring layers in the multi-layer structure devices.

Photoacoustic (PA) spectra on the 3, 4, 9, 10-perylenetetracarboxylic dianhydride (PTCDA) films deposited by the vacuum evaporation were measured. The films have layered structures constructed from the perylene molecule plane structures. The crystal quality depended on the deposited substrate and the photoacoustic spectroscopy (PAS) seems to be the very useful tools to evaluate these properties from the non-radiative features. The films deposited on the three different substrate had the almost same PL spectra, but the films deposited on the glass substrate had the large non-radiative peaks in the PA spectra contrary to the films deposited on the alumina or crystal Si (100) those had the non-radiative peaks only observed at the short wavelength region.

This paper presents a non-scan design scheme to enhance delay fault testability of controllers. In this scheme, we utilize a given state transition graph (STG) to test delay faults in its synthesized controller. The original behavior of the STG is used during test application. For faults that cannot be detected by using the original behavior, we design an extra logic, called an invalid test state and transition generator, to make those faults detectable. Our scheme allows achieving short test application time and at-speed testing. We show the effectiveness of our method by experiments.

This paper describes an approach to feature extraction in speech recognition systems using kernel principal component analysis (KPCA). This approach represents speech features as the projection of the mel-cepstral coefficients mapped into a feature space via a non-linear mapping onto the principal components. The non-linear mapping is implicitly performed using the kernel-trick, which is a useful way of not mapping the input space into a feature space explicitly, making this mapping computationally feasible. It is shown that the application of dynamic (Δ) and acceleration (ΔΔ) coefficients, before and/or after the KPCA feature extraction procedure, is essential in order to obtain higher classification performance. Better results were obtained by using this approach when compared to the standard technique.

The most important function of a router is to perform IP lookup that determines the output ports of incoming IP packets by their destination addresses. Hence, IP lookup is one of the main issues in implementing high-speed routers. The IP lookup algorithm implemented in IQ2200 Chipset with two-level table architecture can efficiently use memory. However, it wastes processor resource for full re-construction of the forwarding tables whenever every route insertion/deletion is requested. In order to improve the utilization of processor resource, we propose an IP lookup algorithm with three-level table architecture for high-speed routers. We evaluate the performance of the proposed algorithm in terms of the memory size required for storing lookup information and the number of memory access in constructing forwarding tables. Being compared with the IQ2200 scheme, the proposed scheme can reduce the number of memory access up to 99% even though it needs about 16% more memory.

This paper proposes a cell layout synthesis method via Boolean Satisfiability (SAT). Cell layout synthesis problems are first transformed into SAT problems by our formulations. Our method realizes a high-speed layout synthesis for CMOS logic cells and guarantees to generate the minimum-width cells with routability under our layout styles. It considers complementary P-/N-MOSFETs individually during transistor placement, and can generate smaller width layout compared with pairing the complementary P-/N-MOSFETs case. To demonstrate the effectiveness of our SAT-based cell synthesis, we present experimental results which compare it with the 0-1 ILP-based transistor placement method and a commercial cell generation tool. The experimental results show that our SAT-based method can generate minimum-width placements in much shorter run time than the 0-1 ILP-based transistor placement method, and can generate the cell layouts of 32 static dual CMOS logic circuits in 54% run time compared with the commercial tool. Area increase of our method without compaction is only 3% compared with the commercial tool with compaction.

This article introduces a novel approach to model phonology and morphosyntax in morpheme unit-based speech recognizers. The proposed methods are evaluated on a Hungarian newspaper dictation task that requires modeling over 1 million different word forms. The architecture of the recognition system is based on the weighted finite-state transducer (WFST) paradigm. The vocabulary units used in the system are morpheme-based in order to provide sufficient coverage of the large number of word-forms resulting from affixation and compounding. Besides the basic pronunciation model and the morpheme N-gram language model we evaluate a novel phonology model and the novel stochastic morphosyntactic language model (SMLM). Thanks to the flexible transducer-based architecture of the system, these new components are integrated seamlessly with the basic modules with no need to modify the decoder itself. We compare the phoneme, morpheme, and word error-rates as well as the sizes of the recognition networks in two configurations. In one configuration we use only the N-gram model while in the other we use the combined model. The proposed stochastic morphosyntactic language model decreases the morpheme error rate by between 1.7 and 7.2% relatively when compared to the baseline trigram system. The proposed phonology model reduced the error rate by 8.32%. The morpheme error-rate of the best configuration is 18% and the best word error-rate is 22.3%.

Borrowing the notion of instantaneous frequency that was developed in the context of time-frequency signal analysis, an instantaneous frequency amplitude spectrum (IFAS) is introduced for estimating fundamental frequency of speech signal in both noiseless and adverse environments. We define harmonicity measure as a quantity that indicates degree of periodical regularity in the IFAS and that shows substantial difference between periodic signal and noise-like waveform. The harmonicity measure is applied to estimate the existence of fundamental frequency. We provide experimental examples to demonstrate the general applicability of the harmonicity measure and apply the proposed procedure to Japanese continuous speech signals. The results show that the proposed method outperforms the conventional methods with or without the presence of noise.

The Hidden Markov Model (HMM) is a popular statistical framework for modeling and analyzing stochastic signals. In this paper, a novel strategy is proposed that makes use of level-building algorithm with a chain of AdaBoost HMM classifiers to model long stochastic processes. AdaBoost HMM classifier belongs to the class of multiple-HMM classifier. It is specially trained to identify samples with erratic distributions. By connecting the AdaBoost HMM classifiers, processes of arbitrary length can be modeled. A probability trellis is created to store the accumulated probabilities, starting frames and indices of each reference model. By backtracking the trellis, a sequence of best-matched AdaBoost HMM classifiers can be decoded. The proposed method is applied to visual speech processing. A selected number of words and phrases are decomposed into sequences of visual speech units using both the proposed strategy and the conventional level-building on HMM method. Experimental results show that the proposed strategy is able to more accurately decompose words/phrases in visual speech than the conventional approach.

LOTOS parallel operator, which is a binary operator, is used to combine processes in order to express their concurrency. Unlike other LOTOS operators, various possibilities exist when combining processes by the parallel operator. If two processes are selected randomly for combining, the size of the composite intermediate process after combining may be large. In this paper, we propose an algorithm for selecting two processes out of three or more processes so that the size of the intermediate process is the smallest when combined by the parallel operator. Smaller size of an intermediate process means it takes less memory space which is very important in designing verification tools for systems or communication protocols specified in LOTOS.

Spectral-amplitude coding (SAC) techniques in fiber-Bragg-grating (FBG)-based optical code-division multiple-access (OCDMA) systems were investigated in our previous work. This paper adopts the same network architecture to investigate the simultaneous reductions of multiple-access interference (MAI) and optical beat interference (OBI). The MAI is caused by overlapping wavelengths from undesired network coder/decoders (codecs) while the OBI is induced by interaction of simultaneous chips at adjacent gratings. It is proposed that MAI and OBI reductions may be obtained by use of: 1) a source spectrum that is divided into equal chip spacing; 2) coded FBGs characterized by approximately the same number of "0" and "1" code elements; and 3) spectrally balanced photo-detectors. With quasi-orthogonal Walsh-Hadamard coded FBGs, complementary spectral chips is employed as signal pairs to be recombined and detected in balanced photo-detectors, thus achieving simultaneous suppression of both MAIs and OBIs. Simulation results showed that the proposed OCDMA spectral-amplitude coding scheme achieves significant MAI and OBI reductions.

This paper is concerned with improving noise-robustness of a multi-SNR multi-band speaker identification system by introducing automatic adjustment of subband likelihood recombination weights. The adjustment is performed on the basis of subband power calculated from the noise observed just before the speech starts in the input signal. To evaluate the noise-robustness of this system, text-independent speaker identification experiments were conducted on speech data corrupted with noises recorded in five environments: "bus," "car," "office," "lobby," and "restaurant". It was found that the present method reduces the identification error by 15.9% compared with the multi-SNR multi-band method with equal recombination weights at 0 dB SNR. The performance of the present method was compared with a clean fullband method in which a speaker model training is performed on clean speech data, and spectral subtraction is applied to the input signal in the speaker identification stage. When the clean fullband method without spectral subtraction is taken as a baseline, the multi-SNR multi-band method with automatic adjustment of recombination weights attained 56.8% error reduction on average, while the average error reduction rate of the clean fullband method with spectral subtraction was 11.4% at 0 dB SNR.

One of the most important issues in spontaneous speech recognition is how to cope with the degradation of recognition accuracy due to speaking rate fluctuation within an utterance. This paper proposes an acoustic model for adjusting mixture weights and transition probabilities of the HMM for each frame according to the local speaking rate. The proposed model is implemented along with variants and conventional models using the Bayesian network framework. The proposed model has a hidden variable representing variation of the "mode" of the speaking rate, and its value controls the parameters of the underlying HMM. Model training and maximum probability assignment of the variables are conducted using the EM/GEM and inference algorithms for the Bayesian networks. Utterances from meetings and lectures are used for evaluation where the Bayesian network-based acoustic models are used to rescore the likelihood of the N-best lists. In the experiments, the proposed model indicated consistently higher performance than conventional HMMs and regression HMMs using the same speaking rate information.

Optimistic coding is a coding in which we require the existence of reliable codes for infinitely many block length. In this letter we consider the optimistic source coding theorems for a general source Z from the information-spectrum approach. We first formulate the problem to be considered clearly. We obtain the optimistic infimum achievable source coding rate Tε (Z) for the case where decoding error probability εn is asymptotically less than or equal to an arbitrarily given ε [0,1). In fact, Tε (Z) turns out to be expressed in a form similar to the ordinary infimum achievable source coding rate. A new expression for Tε (Z) is also given. In addition, we investigate the case where εn = 0 for infinitely many n and obtain the infimum achievable coding rate.

In this paper, a novel carrier-sense multiple-access (CSMA) scheme for UWB ad-hoc network is proposed and evaluated. UWB is a kind of spread spectrum communication and it is possible to detect the distance between the nodes. With this positioning capability of the UWB systems, DS-CDMA (DS-UWB) scheme with variable spreading factor is used. In this paper, a novel CSMA scheme that employs the correlation of the spreading code is proposed.

We analyze the performance of code division multiple access (CDMA) systems with orthogonal frequency division multiplexing (OFDM). We obtain the probability density function (pdf) of the multiple access interference (MAI) of CDMA systems and extend the results to OFDM-CDMA systems to determine the pdf of the MAI and inter-carrier interference (ICI) in terms of the number of users, the spreading length, the crosscorrelation of spreading sequences, the number of sub-carriers and the frequency offset. We consider the synchronous downlink of cellular multi-carrier CDMA systems and derive a Gaussian approximation of the MAI and ICI. The results show that the overall effect of frequency offset varies with system loading for a given crosscorrelation. The performance of OFDM-CDMA in frequency selective fading channels is analyzed in terms of the joint probability of the fading parameter in each sub-carrier.

This paper proposes a new SSB-QPSK modulation/demodulation method. The present method multiplexes the USB (Upper Side Band) and LSB (Lower Side Band) of a QPSK-modulated SSB (Single Side Band) on the same SSB complex frequency band. The present method thus achieves 2 bit/s/Hz. This method is an orthogonal SSB-QPSK method, because the multiplex signals are orthogonal to each other. The demodulator consists of two SSB demodulators. A simulation result in AWGN conditions, shows that the proposed method has better BER (Bit Error Rate) performance than 16 QAM. The degradation of BER in comparison with QPSK is less than 0.2 dB on Eb/No (bit-energy-to-noise-power ratio). In a fading/Doppler environment, the BER performance of the orthogonal SSB-QPSK is the same as that of QPSK.