Precoding techniques can be introduced into relay systems due to the similarity between relay systems and traditional multi-input-multi-output (MIMO) systems. A channel state information feedback scheme is firstly presented for the MIMO relay system in this letter, where the zero-forcing relaying protocol is proposed to be used so that the information of the equivalent channel and the relaying noise can be compressed into two coefficients. With the proposed feedback scheme, the distributed precoding is presented to be applied through two continuous transmitted vectors of the source node while the co-channel interference cancellation equalizer is used in the destination node. The system outage probability can be improved with the precoding in the source node. Furthermore, various spatial data rates can be conveniently supported by the proposed distributed spatial-temporal precoding method.

A scalable speech codec consisting of a harmonic codec as the core layer and MDCT-based transform codec as the enhancement layer is often required to provide both very low-rate core communication and fine granular scalability. This structure, however, has a serious drawback for practical use because a time delay caused by transform in each layer is accumulated, resulting in a long overall codec delay. In this letter, a new MDCT structure is proposed to reduce the overall codec delay by eliminating the accumulation of time delay by each transform. In the proposed structure, the time delay is first reduced by forcing two transforms to share a common look-ahead. The error components of MDCT caused by the look-ahead sharing are then analyzed and compensated in the decoder, resulting in perfect reconstruction. The proposed structure reduces the codec delay by the frame size, with an equivalent coding efficiency.

A 0.9-V 12-bit 40-MSPS pipeline ADC with I/Q amplifier sharing technique is presented for wireless receivers. To achieve high linearity even at 0.9-V supply, the clock signals to sampling switches are boosted over 0.9 V in conversion stages. The clock-boosting circuit for lifting these clocks is shared between I-ch ADC and Q-ch ADC, reducing the area penalty. Low supply voltage narrows the available output range of the operational amplifier. A pseudo-differential (PD) amplifier with two-gain-stage common-mode feedback (CMFB) is proposed in views of its wide output range and power efficiency. This ADC is fabricated in 90-nm CMOS technology. At 40 MS/s, the measured SNDR is 59.3 dB and the corresponding effective number of bits (ENOB) is 9.6. Until Nyquist frequency, the ENOB is kept over 9.3. The ADC dissipates 17.3 mW/ch, whose performances are suitable for ADCs for mobile wireless systems such as WLAN/WiMAX.

Tree structured data often appear in bioinformatics. For example, glycans, RNA secondary structures and phylogenetic trees usually have tree structures. Comparison of trees is one of fundamental tasks in analysis of these data. Various distance measures have been proposed and utilized for comparison of trees, among which extensive studies have been done on tree edit distance. In this paper, we review key results and our recent results on the tree edit distance problem and related problems. In particular, we review polynomial time exact algorithms and more efficient approximation algorithms for the edit distance problem for ordered trees, and approximation algorithms for the largest common sub-tree problem for unordered trees. We also review applications of tree edit distance and its variants to bioinformatics with focusing on comparison of glycan structures.

Traditional wavelet-based speech enhancement algorithms are ineffective in the presence of highly non-stationary noise because of the difficulties in the accurate estimation of the local noise spectrum. In this paper, a simple method of noise estimation employing the use of a voice activity detector is proposed. We can improve the output of a wavelet-based speech enhancement algorithm in the presence of random noise bursts according to the results of VAD decision. The noisy speech is first preprocessed using bark-scale wavelet packet decomposition ( BSWPD ) to convert a noisy signal into wavelet coefficients (WCs). It is found that the VAD using bark-scale spectral entropy, called as BS-Entropy, parameter is superior to other energy-based approach especially in variable noise-level. The wavelet coefficient threshold (WCT) of each subband is then temporally adjusted according to the result of VAD approach. In a speech-dominated frame, the speech is categorized into either a voiced frame or an unvoiced frame. A voiced frame possesses a strong tone-like spectrum in lower subbands, so that the WCs of lower-band must be reserved. On the contrary, the WCT tends to increase in lower-band if the speech is categorized as unvoiced. In a noise-dominated frame, the background noise can be almost completely removed by increasing the WCT. The objective and subjective experimental results are then used to evaluate the proposed system. The experiments show that this algorithm is valid on various noise conditions, especially for color noise and non-stationary noise conditions.

The selection of effective features is especially important in achieving highly accurate speech recognition. Although the mel-cepstrum is a popular and effective feature for speech recognition, it is still unclear that the filterbank adopted in the mel-cepstrum always produces the optimal performance regardless of the phonetic environment of any specific speech recognition task. In this paper, we propose a new cepstral domain feature extraction approach utilizing the entropic distance-based filterbank for highly accurate speech recognition. Experimental results showed that the cepstral features employing the proposed filterbank reduce the relative error by 31% for clean as well as noisy speech compared to the mel-cepstral features.

Substrate noise analysis has become increasingly important in recent LSI design. This is because substrate noise, which affects PLLs, causes jitter that results in timing error. Conventional analysis techniques of substrate noise are, however, impractical for large-scale designs that have hundreds of millions of transistors because the computational complexity is too huge. To solve this problem, we have developed a fast substrate noise analysis technique for large-scale designs, in which a chip is divided into multiple domains and the circuits of each domain are reduced into a macro model. Using this technique, we have designed a processor chip for use in the supercomputer (die size: 20 mm 21 mm, frequency: 3.2 GHz, transistor count: 350M). Computation time with this design is five times faster than that with a 1/3000 scale design using a conventional technique, while resulting discrepancy with measured period jitter is less than 15%.

An orthogonal sequence based MIMO common feedback method for multicast hybrid automatic-repeat-request (H-ARQ) transmission is presented. The proposed method can obtain more diversity gain proportional to the number of transmit antennas than the conventional on-off keying (OOK) based common feedback method. The ACK/NACK detection performance gain of the proposed scheme over the OOK based method is verified by analysis and computer simulation results.

Conventional efficient key recovery attacks against Wired Equivalent Privacy (WEP) require specific initialization vectors or specific packets. Since it takes much time to collect the packets sufficiently, any active attack should be performed. An Intrusion Detection System (IDS), however, will be able to prevent the attack. Since the attack logs are stored at the servers, it is possible to prevent such an attack. This paper proposes an algorithm for recovering a 104-bit WEP key from any IP packets in a realistic environment. This attack needs about 36,500 packets with a success probability 0.5, and the complexity of our attack is equivalent to about 220 computations of the RC4 key setups. Since our attack is passive, it is difficult for both WEP users and administrators to detect our attack.

Dramatic gains in channel capacity can be achieved in the closed-loop MIMO system under the assumption that the base station (BS) can acquire the downlink channel state information (CSI) accurately. However, transmitting CSI with high precision is a heavy burden that wastes a lot of uplink bandwidth, while transmitting CSI within a limited bandwidth leads to the degradation of system performance. To address this problem, we propose a zero-overhead downlink CSI feedback scheme based on the hybrid pilot structure. The downlink CSI is contained in the hybrid pilots at mobile terminal (MT) side, fed back to BS via the uplink pilot channel, and recovered from hybrid pilot at BS side. Meanwhile the uplink channel is estimated based on the hybrid pilot at BS side. Since transmitting the hybrid pilots occupies the same bandwidth as transmitting traditional code division multiplexing based uplink pilots, no extra uplink channel bandwidth is occupied. Therefore, the overhead for downlink CSI feedback is zero. Moreover, the hybrid pilots are formed at MT side by superposing the received analog downlink pilots directly on the uplink pilots. Thus the downlink CSI estimation process is unnecessary at MT side, and MT's complexity can be reduced. Numerical Simulations prove that, the proposed downlink CSI feedback has the higher precision than the traditional feedback schemes while the overhead for downlink CSI feedback is zero.

In this paper, we propose a rapid model adaptation technique for emotional speech recognition which enables us to extract paralinguistic information as well as linguistic information contained in speech signals. This technique is based on style estimation and style adaptation using a multiple-regression HMM (MRHMM). In the MRHMM, the mean parameters of the output probability density function are controlled by a low-dimensional parameter vector, called a style vector, which corresponds to a set of the explanatory variables of the multiple regression. The recognition process consists of two stages. In the first stage, the style vector that represents the emotional expression category and the intensity of its expressiveness for the input speech is estimated on a sentence-by-sentence basis. Next, the acoustic models are adapted using the estimated style vector, and then standard HMM-based speech recognition is performed in the second stage. We assess the performance of the proposed technique in the recognition of simulated emotional speech uttered by both professional narrators and non-professional speakers.

Based on Free Probability Theory (FPT), which has become an important branch of Random Matrix Theory (RMT), a new scheme of frequency band sensing for Cognitive Radio (CR) in Direct-Sequence Code-Division Multiple-Access (DS-CDMA) multiuser network is proposed. Unlike previous studies in the field, the new scheme does not require the knowledge of the spreading sequences of users and is related to the behavior of the asymptotic free behavior of random matrices. Simulation results show that the asymptotic claims hold true even for a small number of observations (which makes it convenient for time-varying topologies) outperforming classical energy detection scheme and another scheme based on random matrix theory.

This letter describes the importance of transition regions, e.g. at phoneme boundaries, for automatic speaker recognition compared with using steady-state regions. Experimental results of automatic speaker identification tasks confirm that transition regions include the most speaker distinctive features. A possible reason for obtaining such results is described in view of articulation, in particular, the degree of freedom of articulators. These results are expected to provide useful information in designing an efficient automatic speaker recognition system.

An analog circuit testing scheme is presented. The testing technique is a sinusoidal fault signature characterization, involving the measurement of DC offset, amplitude, frequency and phase shift, and the realization of two crossing level voltages. The testing system is an extension of the IEEE 1149.4 standard through the modification of an analog boundary module, affording functionalities for both on-chip testing capability, and accessibility to internal components for off-chip testing. A demonstrating circuit-under-test, a 4th-order Gm-C low-pass filter, and the proposed analog testing scheme are implemented in a physical level using 0.18-µm CMOS technology, and simulated using Hspice. Both catastrophic and parametric faults are potentially detectable at the minimum parameter variation of 0.5%. The fault coverage associated with CMOS transconductance operational amplifiers and capacitors are at 94.16% and 100%, respectively. This work offers the enhancement of standardizing test approach, which reduces the complexity of testing circuit and provides non-intrusive analog circuit testing.

We present a BPSK coherent optical wireless link in a multiple-beam, multiple-aperture configuration. The data are recovered using the signal obtained by the coherent addition of a set of maximum likelihood optical phase estimates and a select-largest stage. The proposal offers higher performance than the combining methods commonly used in optical wireless systems with diversity transmission and coherent detection.

This paper presents a single-chip speech dialogue module and its evaluation on a personal robot. This module is implemented on an application processor that was developed primarily for mobile phones to provide a compact size, low power-consumption, and low cost. It performs speech recognition with preprocessing functions such as direction-of-arrival (DOA) estimation, noise cancellation, beamforming with an array of microphones, and echo cancellation. Text-to-speech (TTS) conversion is also equipped with. Evaluation results obtained on a new personal robot, PaPeRo-mini, which is a scale-down version of PaPeRo, demonstrate an 85% correct rate in DOA estimation, and as much as 54% and 30% higher speech recognition rates in noisy environments and during robot utterances, respectively. These results are shown to be comparable to those obtained by PaPeRo.

A circuit technique to correct Vdd/PM distortion and improve efficiency as supply modulation of cascode class-E PAs has been proposed. The experimental result shows that the phase distortion can be improved from 20 degrees to 5 degrees. Moreover, a system co-simulation result demonstrated that the EVM can be improved from -17 dB to -19 dB.

This paper studies scattering and diffraction of a TE plane wave from a periodic surface with semi-infinite extent. By use of a combination of the Wiener-Hopf technique and a perturbation method, a concrete representation of the wavefield is explicitly obtained in terms of a sum of two types of Fourier integrals. It is then found that effects of surface roughness mainly appear on the illuminated side, but weakly on the shadow side. Moreover, ripples on the angular distribution of the first-order scattering in the shadow side are newly found as interference between a cylindrical wave radiated from the edge and an inhomogeneous plane wave supported by the periodic surface.

Through the analysis of Rayleigh model, an explanatory model for the quality effect of peer reviews is constructed. The review activities are evaluated by the defect removal rate at each phase. We made hypotheses on how these measurements are related to the product quality. These hypotheses are verified through regression analysis of actual project data, and concrete calculation formulae are obtained as a model. Making use of the mechanism to construct this model, we can develop a method for making concrete review plan and setting objective values to manage on-going review activities.

Wireless mesh networks have been attracting many users in recent years. By connecting base stations (mesh nodes) with wireless connections, these network can achieve a wide-area wireless environment with flexible configuration and low cost at the risk of radio interference between wireless links. When we utilize wireless mesh networks as infrastructures for Internet access, all network traffic from mobile nodes goes through a gateway node that is directly connected to the wired network. Therefore, it is necessary to distribute the traffic load by deploying multiple gateway nodes. In this paper, we propose a spanning tree construction algorithm for TDMA-based wireless mesh networks with multiple gateway nodes so as to maximize the traffic volume transferred between the mesh network and the Internet (system throughput) by taking account of the traffic load on the gateway nodes, the access link capacity and radio interference. Through a performance evaluation, we show that the proposed algorithm increases the system throughput regardless of the bottleneck position and achieves up to 3.1 times higher system throughput than a conventional algorithm.