A 1.8 V, 5 GHz low power frequency synthesizer for Wireless Sensor Networks is presented in 0.18 µm CMOS technology. A low power phase-switching prescaler is designed, and the current mode phase rotator is merged into the first divide-by-2 circuit of the prescaler to reduce power and propagation delay. An improved charge pump circuit is proposed to compensate for the dynamic effects with the charge pump. By a divide-by-2 circuit, the frequency synthesizer can provide a 2.324-2.714 GHz quadrature output frequency in 1 MHz steps with a 4 MHz reference frequency. The measured output phase noise is -110 dBc/Hz at 1-MHz offset frequency. The power consumption of the PLL is 11.2 mW at 1.8 V supply voltage.

Effect of an addition of a cooled step-up transformer to a flux locked loop (FLL) circuit was studied to reduce indirect rf interference to HTS-dc-SQUID. First, we demonstrated that a noise level of an HTS-dc-SQUID system using the FLL circuit with single room-temperature transformer could be easily degraded by radiation of rf electromagnetic wave to cables in the FLL circuit. It is thought that the rf radiation induced rf current in the circuit, and was transmitted to the SQUID to modulate the bias current, resulting in the increase of the noise level. To avoid the degradation due to such indirect rf interference, the cooled set-up transformer was added to the FLL circuit since it was expected that the additional transformer would work as a "step-down" transformer against the induced rf current. It was shown that the noise level of a HTS-SQUID system (SQUITEM system) operated in an electromagnetically unshielded environment could be improved to the same level as that measured in a magnetically shielded room by the additional cooled transformer and appropriate impedance matching.

We propose a neutron diffractometer system based on MgB2 thin film detectors and an SFQ signal processor. Small dimensions of MgB2 thin film detectors and high processing capability of the single flux quantum (SFQ) circuits enable us to handle several thousand or more detectors in a cryocooler, leading to a very compact system. In addition, the system can provide many diffraction patterns for different kinetic energies simultaneously. Kinetic energy is determined for individual neutrons by means of the time-of-flight method by using SFQ time-to-digital converters (TDCs). Digital outputs of the TDCs are multiplexed in time domain and sent to room-temperature electronics with reduced number of cables. A dual-input SFQ signal processor including TDCs and a multiplexer has been successfully demonstrated with a time resolution of 20 ns and power consumption of 400 µW. These values show high feasibility of the neutron diffraction system proposed here.

In this paper, we propose a method for layout-driven skewed clock tree synthesis for SFQ logic circuits. For a given logic circuit without a clock tree, our algorithm outputs a circuit with a synthesized clock tree and timing adjustments achieving the given clock period and a rough placement of the clocked gates. In the proposed algorithm, clocked gates are grouped into levels and the clock tree is synthesized for each level. For each level, we estimate the clock timing for all possible placements of each gate, and then we search a placement of all gates that minimizes the total number of delay elements for timing adjustment. Once the placement is obtained, we synthesize a clock tree without wire intersections. We applied the proposed method to a moderate size circuit and confirmed that clock trees satisfying given timing requirements can be synthesized automatically.

Although larger scale integration enhances the practicability of superconducting Josephson circuits, several technical problems begin to emerge during its progress. One of the problems is the increase of current through a ground plane (ground current). Excess ground current produces additional magnetic field and reduces operation margins of the circuits, because superconducting Josephson devices are very sensitive to magnetic field. In this paper, we evaluate current distribution in a superconducting ground plane by means of both experiments and numerical calculation. We also verify two methods for suppressing the ground current. One is a slot structure in the ground plane, and the other is alignment of the current-extraction point. Suppression of the ground current is quantitatively evaluated.

This paper proposes a modified per-survivor iterative timing recovery scheme, which exploits a new split-preamble strategy in conjunction with a per-survivor processing soft-output Viterbi algorithm (PSP-SOVA). The conventional split-preamble strategy places a preamble at the beginning of a data sector and uses it to run a phase-locked loop during acquisition to find an initial phase/frequency offset. However, the proposed scheme splits the preamble into two parts. The first part is placed at the beginning of the data sector, whereas the second part is divided into small clusters, each of which is then embedded uniformly within the data stream. This split preamble is utilized to adjust the branch metric calculation in PSP-SOVA to ensure that the survivor path occurs in a correct direction. Results indicate that the proposed scheme yields a better performance than a conventional receiver with separate timing recovery and turbo equalization, and the iterative timing recovery scheme proposed in [1],[2], especially when the timing jitter is large. In addition, we also show that the proposed scheme can automatically correct a cycle slip much more efficiently than the others.

A new selective retransmission method for HARQ (Hybrid Automatic Repeat reQuest) systems is proposed. This method can avoid the blindness of symbols retransmission by the transformation of lostmap matrix and simulation results show that the proposed method can reduce the number of retransmissions effectively.

In this letter, we give a generalized construction for sets of frequency-hopping sequences (FHSs) based on power-residue sequences. Our construction encompasses a known optimal construction and can generate new optimal sets of FHSs which simultaneously achieve the Peng-Fan bound and the Lempel-Greenberger bound.

A new state estimation algorithm is presented for a class of LTI systems that have an input disturbance in polynomial form and a sinusoidal sensor disturbance in the measurement output. Adaptation rules are developed for identifying the unknown magnitude, phase and frequency of the sensor disturbance from the system output measurement. For the application of the identification result to the state estimation problem, the sinusoidal signal with arbitrary initial phase has been considered in this paper. In order to test the performance of the proposed algorithm, comparative computer simulations have been carried out with a robust state observer. Simulation results show the effectiveness of the proposed method.

We propose a blind dereverberation method based on spectral subtraction using a multi-channel least mean squares (MCLMS) algorithm for distant-talking speech recognition. In a distant-talking environment, the channel impulse response is longer than the short-term spectral analysis window. By treating the late reverberation as additive noise, a noise reduction technique based on spectral subtraction was proposed to estimate the power spectrum of the clean speech using power spectra of the distorted speech and the unknown impulse responses. To estimate the power spectra of the impulse responses, a variable step-size unconstrained MCLMS (VSS-UMCLMS) algorithm for identifying the impulse responses in a time domain is extended to a frequency domain. To reduce the effect of the estimation error of the channel impulse response, we normalize the early reverberation by cepstral mean normalization (CMN) instead of spectral subtraction using the estimated impulse response. Furthermore, our proposed method is combined with conventional delay-and-sum beamforming. We conducted recognition experiments on a distorted speech signal simulated by convolving multi-channel impulse responses with clean speech. The proposed method achieved a relative error reduction rate of 22.4% in relation to conventional CMN. By combining the proposed method with beamforming, a relative error reduction rate of 24.5% in relation to the conventional CMN with beamforming was achieved using only an isolated word (with duration of about 0.6 s) to estimate the spectrum of the impulse response.

The increasing demand of low power Direct Digital Frequency Synthesizer (DDFS) leads to the requirement of efficient compression methods to reduce ROM size for storing sine function values. This paper presents a technique to achieve very high compression ratio by using the optimized four-segment linear difference method. The proposed technique results in the ROM compression ratio of about 117.3:1 and the word size reduction of 6 bits for the design of a DDFS with 11-bit sine amplitude output. This high compression ratio result is very promising to meet the requirement of low power consumption and low hardware complexity in digital VLSI technology.

We have designed and implemented a 4-bit Carry Look-ahead Adder (CLA) and 4-bit parallel multipliers to be used for the Fast Fourier Transform (FFT) system with the estimated clock frequency of 20 GHz. Through some high frequency functional tests, we have confirmed that the operation of the CLA has been successful. Through some low speed tests, we have also confirmed that the operation of multiplication has been successful. In addition, we have designed a 4-bit multiplier with a Booth encoder and with a 2-point-4-bit butterfly circuit.

XQuery has become the standard for querying XML. Just like SQL, XQuery allows nested expressions. To optimize XQuery processing, a lot of research has been done on normalization, i.e., transforming nested expressions to equivalent unnested ones. Previous normalization rules are classified into two categories—source-level/ and algebra-level/—depending on whether a construct is specified in the XQuery syntax or as equivalent algebraic expressions. From an implementation point of view, the former is preferable to the latter since it can be implemented in a variety of XQuery engines with different algebras. However, existing source-level rules have several problems: They do not handle quantified expressions, incur duplicated query results, and use many temporary files. In this paper, we propose new source-level normalization rules that solve these problems. Through analysis and experiments, we show that our normalization rules can reduce query execution time from hours to a few seconds and can be adapted to a variety of XQuery engines.

In this paper, we introduce a new sequential pattern, the Interactive User Sequence Pattern (IUSP). This pattern is useful for grouping highly interrelated users in one-way communications such as e-mail, SMS, etc., especially when the communications include many spam users. Also, we propose an efficient algorithm for discovering IUSPs from massive one-way communication logs containing only the following information: senders, receivers, and dates and times. Even though there is a difficulty in that our new sequential pattern violates the Apriori property, the proposed algorithm shows excellent processing performance and low storage cost in experiments on a real dataset.

This paper proposes an efficient combination of state likelihood recycling and batch state likelihood calculation for accelerating acoustic likelihood calculation in an HMM-based speech recognizer. Recycling and batch calculation are each based on different technical approaches, i.e. the former is a purely algorithmic technique while the latter fully exploits computer architecture. To accelerate the recognition process further by combining them efficiently, we introduce conditional fast processing and acoustic backing-off. Conditional fast processing is based on two criteria. The first potential activity criterion is used to control not only the recycling of state likelihoods at the current frame but also the precalculation of state likelihoods for several succeeding frames. The second reliability criterion and acoustic backing-off are used to control the choice of recycled or batch calculated state likelihoods when they are contradictory in the combination and to prevent word accuracies from degrading. Large vocabulary spontaneous speech recognition experiments using four different CPU machines under two environmental conditions showed that, compared with the baseline recognizer, recycling and batch calculation, our combined acceleration technique further reduced both of the acoustic likelihood calculation time and the total recognition time. We also performed detailed analyses to reveal each technique's acceleration and environmental dependency mechanisms by classifying types of state likelihoods and counting each of them. The analysis results comfirmed the effectiveness of the combined acceleration technique.

In this paper, we establish our child safety system model related to the addressing contradictory issue of wireless sensor networks caused by the mutual authentication and privacy protection of an end-user. Based on the system model, we propose the novel location-aware and privacy-preserving approach for providing child safety over wireless sensor networks. Although we illustrate our protocol over the sensor networks, the proposed protocol can be operated by various wireless networks (e.g., WiFi and UWB) which can support RSSI (Received Signal Strength Indication). Compared to a few previous works, the proposed approach can show the potential of enhancing accuracy with location information, preserve the privacy of an end-user, and provide the capability of controlling the child safety service to an end-user.

Providing mobile services that ensure user satisfaction is one of the most crucial issues for telecommunication carriers and service providers. Traditionally, user satisfaction with the service, i.e., the Quality of Experience (QoE), has been assessed in a laboratory by using elaborate network systems and customized terminals. However, reliable QoE for a target mobile service in the context of actual use cannot be obtained by laboratory experiment, since QoE can be affected by a variety of factors such as environmental conditions and the mental state of the user. This paper proposes a novel Web script-based field evaluation method designed to ascertain user satisfaction with mobile services. The proposed method has the following practical advantages. Since experimental conditions, especially communication conditions, can be simulated with a Web script, such as FlashLite, a subjective assessment can be conducted anywhere using the mobile terminal of the participant. Thus, QoE for a target mobile service in the field can easily be obtained.

Recently, overlap frequency-domain equalization (FDE) based on minimum mean square error (MMSE) criterion which requires no guard interval (GI) insertion was proposed for signal transmission using multi-code direct sequence code division multiple access (DS-CDMA) in a frequency-selective fading channel. Another promising equalization technique is time-domain sliding-window chip equalization (SWCE). In this paper, the bit error rate (BER) performances achievable with overlap FDE and SWCE are compared. It is shown that, by extending the fast Fourier transform (FFT) block size, overlap FDE can achieve almost the same BER performance as SWCE with much less computational complexity than SWCE.

One typical phonotactic system for language recognition is parallel phone recognition followed by vector space modeling (PPRVSM). In this system, various phone recognizers are applied in parallel and fused at the score level. Each phone recognizer is trained for a known language, which is assumed to extract complementary information for effective fusion. But this method is limited by the large amount of training samples for which word or phone level transcription is required. Also, score fusion is not the optimal method as fusion at the feature or model level will retain more information than at the score level. This paper presents a new strategy to build and fuse parallel phone recognizers (PPR). This is achieved by training multiple acoustic diversified phone recognizers and fusing at the feature level. The phone recognizers are trained on the same speech data but using different acoustic features and model training techniques. For the acoustic features, Mel-frequency cepstral coefficients (MFCC) and perceptual linear prediction (PLP) are both employed. In addition, a new time-frequency cepstrum (TFC) feature is proposed to extract complementary acoustic information. For the model training, we examine the use of the maximum likelihood and feature minimum phone error methods to train complementary acoustic models. In this study, we fuse phonotactic features of the acoustic diversified phone recognizers using a simple linear fusion method to build the PPRVSM system. A novel logistic regression optimized weighting (LROW) approach is introduced for fusion factor optimization. The experimental results show that fusion at the feature level is more effective than at the score level. And the proposed system is competitive with the traditional PPRVSM. Finally, the two systems are combined for further improvement. The best performing system reported in this paper achieves an equal error rate (EER) of 1.24%, 4.98% and 14.96% on the NIST 2007 LRE 30-second, 10-second and 3-second evaluation databases, respectively, for the closed-set test condition.

We report on the enhancement of system detection efficiency in a superconducting nanowire single-photon detector (SNSPD) by applying the optical cavity structure. The nanowire was made using 4-nm-thick NbN thin films and covered with an SiO cavity and Au mirror designed for 1300-1600 nm wavelengths. The device is mounted into fiber-coupled packages, and installed in a practical multichannel system based on GM cryocoolers. System detection efficiency depends on the absorptance of cavity structure, and reached 28% and 40% at 1550 nm and 1310 nm wavelengths, respectively. These values were considerably higher than an SNSPD without optical cavity.