We have designed a superconductor-semiconductor hybrid analog-to-digital (A/D) converter and experimentally evaluated its performance at sampling frequencies up to 18.6 GHz. The A/D converter consists of a superconductor front-end circuit and a semiconductor back-end circuit. The front-end circuit includes a sigma-delta modulator and an interface circuit, which is for transmitting data signal to the semiconductor back-end circuit. The semiconductor back-end circuit performs decimation filtering. The design of the modulator was modified to reduce effects of integrator leak and thermal noise on signal-to-noise ratio (SNR). Using the improved modulator design, we achieved a bit-accuracy close to the ideal value. The hybrid architecture enabled us to reduce the integration scale of the front-end circuit to fewer than 500 junctions. This simplicity makes feasible a circuit based on a high TC superconductor as well as on a low TC superconductor. The experimental results show that the hybrid A/D converter operated perfectly and that SNR was 84.8 dB (bit accuracy~13.8 bit) at a band width of 9.1 MHz. This converter has the highest performance of all sigma-delta A/D converters.

This paper addresses the problem of voice activity detection (VAD) in noisy environments. The VAD method proposed in this paper is based on a statistical model approach, and estimates statistical models sequentially without a priori knowledge of noise. Namely, the proposed method constructs a clean speech / silence state transition model beforehand, and sequentially adapts the model to the noisy environment by using a switching Kalman filter when a signal is observed. In this paper, we carried out two evaluations. In the first, we observed that the proposed method significantly outperforms conventional methods as regards voice activity detection accuracy in simulated noise environments. Second, we evaluated the proposed method on a VAD evaluation framework, CENSREC-1-C. The evaluation results revealed that the proposed method significantly outperforms the baseline results of CENSREC-1-C as regards VAD accuracy in real environments. In addition, we confirmed that the proposed method helps to improve the accuracy of concatenated speech recognition in real environments.

In recent decades, there has been a great deal of research into the problem of bilingual speech recognition - to develop a recognizer that can handle inter- and intra-sentential language switching between two languages. This paper presents our recent work on the development of a grammar-constrained, Mandarin-English bilingual Speech Recognition System (MESRS) for real world music retrieval. Two of the main difficult issues in handling the bilingual speech recognition systems for real world applications are tackled in this paper. One is to balance the performance and the complexity of the bilingual speech recognition system; the other is to effectively deal with the matrix language accents in embedded language. In order to process the intra-sentential language switching and reduce the amount of data required to robustly estimate statistical models, a compact single set of bilingual acoustic models derived by phone set merging and clustering is developed instead of using two separate monolingual models for each language. In our study, a novel Two-pass phone clustering method based on Confusion Matrix (TCM) is presented and compared with the log-likelihood measure method. Experiments testify that TCM can achieve better performance. Since potential system users' native language is Mandarin which is regarded as a matrix language in our application, their pronunciations of English as the embedded language usually contain Mandarin accents. In order to deal with the matrix language accents in embedded language, different non-native adaptation approaches are investigated. Experiments show that model retraining method outperforms the other common adaptation methods such as Maximum A Posteriori (MAP). With the effective incorporation of approaches on phone clustering and non-native adaptation, the Phrase Error Rate (PER) of MESRS for English utterances was reduced by 24.47% relatively compared to the baseline monolingual English system while the PER on Mandarin utterances was comparable to that of the baseline monolingual Mandarin system. The performance for bilingual utterances achieved 22.37% relative PER reduction.

A new adaptive filter algorithm based on the linear prediction property of sinusoidal signals is proposed for unbiased estimation of the frequency of a real tone in white noise. Similar to the least mean square algorithm, the estimator is computationally simple and it provides unbiased as well as direct frequency measurements. Learning behavior and variance of the estimated frequency are derived and confirmed by computer simulations.

This letter presents a race-free mixed serial-parallel comparison (RFMSPC) scheme which uses both serial and parallel CAMs in a match line. A self-reset search line scheme for the serial CAM is proposed to avoid the timing race problem and additional timing penalties. Various 32 entry CAMs are designed using 90 nm 1.2 V CMOS process to verify the proposed RFMSPC scheme. It shows that the RFMSPC saves power consumption by 40%, 53% and 63% at the cost of a 4%, 6% and 16% increase in search time according to 1, 2, and 4 serial CAM bits in a match line.

In this paper, a block-constrained trellis coded quantization (BC-TCQ) algorithm is combined with an algebraic codebook to produce an algebraic trellis code (ATC) to be used in ACELP coding. In ATC, the set of allowed algebraic codebook pulse positions is expanded, and the expanded set is partitioned into subsets of pulse positions; the trellis branches are labeled with these subsets. The list Viterbi algorithm (LVA) is used to select the excitation codevector. The combination of an ATC codebook and LVA trellis search algorithm is denoted as an ATC-LVA block code. The ATC-LVA block code is used as the fixed codebook of the AMR-WB 8.85 kbps mode, reducing complexity compared to the conventional algebraic codebook.

This paper presents methods for detecting transistor short faults using logic level fault simulation and test generation. The paper considers two types of transistor level faults, namely strong shorts and weak shorts, which were introduced in our previous research. These faults are defined based on the values of outputs of faulty gates. The proposed fault simulation and test generation are performed using gate-level tools designed to deal with stuck-at faults, and no transistor-level tools are required. In the test generation process, a circuit is modified by inserting inverters, and a stuck-at test generator is used. The modification of a circuit does not mean a design-for-testability technique, as the modified circuit is used only during the test generation process. Further, generated test patterns are compacted by fault simulation. Also, since the weak short model involves uncertainty in its behavior, we define fault coverage and fault efficiency in three different way, namely, optimistic, pessimistic and probabilistic and assess them. Finally, experimental results for ISCAS benchmark circuits are used to demonstrate the effectiveness of the proposed methods.

This study deals with two linear precoders: the maximization of the minimum Euclidean distance between received symbol-vectors, called here max-dmin, and the maximization of the post-processing signal-to-noise ratio termed max-SNR or beamforming. Both have been designed for reliable MIMO transmissions operating over uncorrelated Rayleigh fading channels. Here, we will explain why performances in terms of bit error rates show a significant enhancement of the max-dmin over the max-SNR whenever the number of antennas is increased. Then, from theoretical developments, we will demonstrate that, like the max-SNR precoder, the max-dmin precoder achieves the maximum diversity order, which is warrant of reliable transmissions. The current theoretical knowledge will be applied to the case-study of a system with two transmit- or two receive-antennas to calculate the probability density functions of two channel parameters directly linked to precoder performances for uncorrelated Rayleigh fading channels. At last, this calculation will allow us to quickly get the BER of the max-dmin precoder further to the derivation of a tight semi-theoretical approximation.

This paper presents a target location estimation method that can use a pair of leaky coaxial cables to determine the 2D coordinates of the target. Since convention location techniques using leaky coaxial cables can find a target location along the cable in 1D, they have been unable to locate it in 2D planes. The proposed method enables us to estimate target on a 2D plane using; 1) a beam-forming technique and 2) a reconstruction technique based on Hough transform. Leaky coaxial cables are equipped with numerous slots at regular interval, which can be utilized as antenna arrays that acts both as transmitters and receivers. By completely exploiting this specific characteristic of leaky coaxial cables, we carried out an antenna array analysis that performs in a beam-forming fashion. Simulation and experimental results support the effectiveness of the proposed target location method.

We consider a data set in which each example is an n-dimensional Boolean vector labeled as true or false. A pattern is a co-occurrence of a particular value combination of a given subset of the variables. If a pattern appears frequently in the true examples and infrequently in the false examples, we consider it a good pattern. In this paper, we discuss the problem of determining the data size needed for removing "deceptive" good patterns; in a data set of a small size, many good patterns may appear superficially, simply by chance, independently of the underlying structure. Our hypothesis is that, in order to remove such deceptive good patterns, the data set should contain a greater number of examples than that at which a random data set contains few good patterns. We justify this hypothesis by computational studies. We also derive a theoretical upper bound on the needed data size in view of our hypothesis.

In order to achieve the prioritized quality of service (QoS) guarantee, the IEEE 802.11e EDCAF (the enhanced distributed channel access function) provides the distinguished services by configuring the different QoS parameters to different access categories (ACs). An admission control scheme is needed to maximize the utilization of wireless channel. Most of papers study throughput improvement by solving the complicated multidimensional Markov-chain model. In this paper, we introduce a backoff model to study the transmission probability of the different arbitration interframe space number (AIFSN) and the minimum contention window size (CWmin). We propose an adaptive control scheme (ACS) to dynamically update AIFSN and CWmin based on the periodical monitoring of current channel status and QoS requirements to achieve the specific service differentiation at access points (AP). This paper provides an effective tuning mechanism for improving QoS in WLAN. Analytical and simulation results show that the proposed scheme outperforms the basic EDCAF in terms of throughput and service differentiation especially at high collision rate.

In this paper, we propose a novel sound field reproduction system that uses the simultaneous perturbation (SP) method as well as two fast convergence techniques. Sound field reproduction systems that reproduce any desired signal at listener's ear generally use fixed preprocessing filters that are determined by the transfer functions from loudspeakers to control points in advance. However, control point movement results in severe localization errors. Our solution is a sound field reproduction system, based on the SP method, which uses only an error signal to update the filter coefficients. The SP method can track all control point movements but suffers from slow convergence. Hence, we also propose two methods that offer improved convergence speeds. One is a delay control method that compensates the delay caused by back-and-forth control point movements. The other is a compensation method that offsets the localization error caused by head rotation. Simulations demonstrate that the proposed methods can well track control point movements while offering reasonable convergence speeds.

A lightweight and high gain planar antenna for space use is realized with radial waveguide slotted array and honeycomb structure with the weight of 1.16 kg and the diameter of 920.5 mm. The slot coupling is analyzed by method of moments considering the hybrid mode in the multi-layer waveguide structure. The propagation constant of the honeycomb structure is measured and the low-loss property is obtained at the frequency range of 8 GHz. The fabricated RLSA is measured and the reflection is around -10 dB in 8 GHz band. The measured aperture fields agree with the calculation in the radial direction. In the azimuthal direction, on the other hand, the fields show ripples of 6 dB and 60 degree. The gain of 35.9 dBi with the efficiency of 58.7% is obtained at 8.6 GHz.

An efficient way to develop large scale speech corpora is to collect phonetically rich ones that have high coverage of phonetic contextual units. The sentence set, usually called as the minimum set, should have small text size in order to reduce the collection cost. It can be selected by a greedy search algorithm from a large mother text corpus. With the inclusion of more and more phonetic contextual effects, the number of different phonetic contextual units increased dramatically, making the search not a trivial issue. In order to improve the search efficiency, we previously proposed a so-called least-to-most-ordered greedy search based on the conventional algorithms. This paper evaluated these algorithms in order to show their different characteristics. The experimental results showed that the least-to-most-ordered methods successfully achieved smaller objective sets at significantly less computation time, when compared with the conventional ones. This algorithm has already been applied to the development a number of speech corpora, including a large scale phonetically rich Chinese speech corpus ATRPTH which played an important role in developing our multi-language translation system.

The author developed a GaAs wideband IQ modulator IC, which is utilized in RF signal source instruments with direct-conversion architecture. The layout is fully symmetric to obtain a temperature-stable operation. However, the actual temperature drift of EVM (Error Vector Magnitude) is greater in some frequency and temperature ranges than the first generation IC of the same architecture. For applications requiring the precision of electric instrumentation, temperature drift is highly critical. This paper clarifies that linear phase error is the dominant factor causing the temperature drift. It also identifies that such temperature drift of linear phase error is due to equivalent series impedance, especially parasitic capacitance of the phase shifter. This effect is verified by comparing the SSB measurements to a mathematical simulation using an empirical temperature-dependent small-signal FET model.

We present a novel method of integrating the likelihoods of multiple feature streams, representing different acoustic aspects, for robust speech recognition. The integration algorithm dynamically calculates a frame-wise stream weight so that a higher weight is given to a stream that is robust to a variety of noisy environments or speaking styles. Such a robust stream is expected to show discriminative ability. A conventional method proposed for the recognition of spoken digits calculates the weights from the entropy of the whole set of HMM states. This paper extends the dynamic weighting to a real-time large-vocabulary continuous speech recognition (LVCSR) system. The proposed weight is calculated in real-time from mutual information between an input stream and active HMM states in a search space without an additional likelihood calculation. Furthermore, the mutual information takes the width of the search space into account by calculating the marginal entropy from the number of active states. In this paper, we integrate three features that are extracted through auditory filters by taking into account the human auditory system's ability to extract amplitude and frequency modulations. Due to this, features representing energy, amplitude drift, and resonant frequency drifts, are integrated. These features are expected to provide complementary clues for speech recognition. Speech recognition experiments on field reports and spontaneous commentary from Japanese broadcast news showed that the proposed method reduced error words by 9.2% in field reports and 4.7% in spontaneous commentaries relative to the best result obtained from a single stream.

The depth-first sphere decoder (SD) and the K-best algorithm have been widely studied as near optimum detectors. Depth-first SD has a non-deterministic computational throughput and K-best requires a sorting unit whose complexity is significant when a large K is used together with high modulation constellations. In this letter, we propose a MIMO detector that employs the trellis structure instead of the conventional tree searching. This detector can keep the computational throughput constant and reduce the complexity because the sorting is not required. From the simulation and complexity analysis, we investigate the advantage and drawback of the proposed detector.

In forensic voice telephony speaker verification, we may be requested to identify a speaker in a very noisy environment, unlike the conditions in general research. In a noisy environment, we process speech first by clarifying it. However, the previous study of speaker verification from clarified speech did not yield satisfactory results. In this study, we experimented on speaker verification with clarification of speech in a noisy environment, and we examined the relationship between improving acoustic quality and speaker verification results. Moreover, experiments with realistic noise such as a crime prevention alarm and power supply noise was conducted, and speaker verification accuracy in a realistic environment was examined. We confirmed the validity of speaker verification with clarification of speech in a realistic noisy environment.

Realistic representations using extremely high quality images are becoming increasingly popular. For example, digital cinemas can now display moving pictures composed of high-resolution digital images. Although these applications focus on increasing the spatial resolution only, higher frame-rates are being considered to achieve more realistic representations. Since increasing the frame-rate increases the total amount of information, efficient coding methods are required. However, its statistical properties are not clarified. This paper establishes for high frame-rate video a mathematical model of the relationship between frame-rate and bit-rate. A coding experiment confirms the validity of the mathematical model.

We have developed the fabrication process, the circuit design technology, and the cryopackaging technology for high-Tc single flux quantum (SFQ) devices with the aim of application to an analog-to-digital (A/D) converter circuit for future wireless communication and a sampler system for high-speed measurements. Reproducibility of fabricating ramp-edge Josephson junctions with IcRn products above 1 mV at 40 K and small Ic spreads on a superconducting groundplane was much improved by employing smooth multilayer structures and optimizing the junction fabrication process. The separated base-electrode layout (SBL) method that suppresses the Jc spread for interface-modified junctions in circuits was developed. This method enabled low-frequency logic operations of various elementary SFQ circuits with relatively wide bias current margins and operation of a toggle-flip-flop (T-FF) above 200 GHz at 40 K. Operation of a 1:2 demultiplexer, one of main elements of a hybrid-type Σ-Δ A/D converter circuit, was also demonstrated. We developed a sampler system in which a sampler circuit with a potential bandwidth over 100 GHz was cooled by a compact stirling cooler, and waveform observation experiments confirmed the actual system bandwidth well over 50 GHz.