In this paper we propose a cross-layer retransmission control for TCP communication over a wireless link. With our proposed control, a retransmission delay for lost packet is reduced, packet losses in the wireless link are eliminated and all packets are delivered in the correct order. No change is required to TCP itself or to the sender. Our proposed method is implemented in a queue between the media access control (MAC) layer and logical link layer in a base station, and is designed to assist local retransmission control in the MAC layer. Computer simulations show that our proposed method can maximally use the bandwidth of the wireless link under high bit error rates conditions with conventional TCP control. The fairness problem of TCP communication between connections with different bit error rates in a wireless link is also improved, and MAC level fairness is also controllable.

This paper considers reduction of the peak-to-average power ratio (PAPR) of M-quadrature amplitude modulation (QAM) signals in orthogonal frequency division multiplexing (OFDM) systems. It is known that a 16-QAM or 64-QAM constellation can be written as the vector sum of two or three QPSK constellations respectively. We can then use the Golay complementary sequences over Z4 to construct 16-QAM or 64-QAM OFDM sequences with low PAPR. In this paper, we further examine the squared Euclidean distance of these M-QAM sequences and their variations. Our goal here is to combine the block coded modulation (BCM) and Golay complementary sequences to trade off the PAPR, the code rate, and the squared Euclidean distance of M-QAM OFDM signals. In particular, some 16-QAM and 64-QAM OFDM sequences with low PAPR and large squared Euclidean distance are presented.

This paper describes a new method to detect speech segments online with identifying gender attributes for efficient dual gender-dependent speech recognition and broadcast news captioning. The proposed online speech detection performs dual-gender phoneme recognition and detects a start-point and an end-point based on the ratio between the cumulative phoneme likelihood and the cumulative non-speech likelihood with a very small delay from the audio input. Obtaining the speech segments, the phoneme recognizer also identifies gender attributes with high discrimination in order to guide the subsequent dual-gender continuous speech recognizer efficiently. As soon as the start-point is detected, the continuous speech recognizer with paralleled gender-dependent acoustic models starts a search and allows search transitions between male and female in a speech segment based on the gender attributes. Speech recognition experiments on conversational commentaries and field reporting from Japanese broadcast news showed that the proposed speech detection method was effective in reducing the false rejection rate from 4.6% to 0.53% and also recognition errors in comparison with a conventional method using adaptive energy thresholds. It was also effective in identifying the gender attributes, whose correct rate was 99.7% of words. With the new speech detection and the gender identification, the proposed dual-gender speech recognition significantly reduced the word error rate by 11.2% relative to a conventional gender-independent system, while keeping the computational cost feasible for real-time operation.

This paper presents a novel approach to single channel speech enhancement in noisy environments. Widely adopted noise reduction techniques based on the spectral subtraction are generally expressed as a spectral gain depending on the signal-to-noise ratio (SNR) [1]-[4]. As the estimation method of the SNR, the well-known decision-directed (DD) estimator of Ephraim and Malah efficiently is known to reduces musical noise in noise frames, but the a priori SNR, which is a crucial parameter of the spectral gain, follows the a posteriori SNR with a delay of one frame in speech frames [5]. Therefore, the noise suppression gain using the delayed a priori SNR, which is estimated by the DD algorithm matches the previous frame rather than the current one, so after noise suppression, this degrades the performance of a noise reduction during abrupt transient parts. To overcome this artifact, we propose a computationally simple but effective speech enhancement technique based on the sigmoid type function to adaptively determine the weighting factor of the DD algorithm. Actually, the proposed approach avoids the delay problem of the a priori SNR while maintaining the advantage of the DD algorithm. The performance of the proposed enhancement algorithm is evaluated by the objective and subjective test under various environments and yields better results compared with the conventional DD scheme based approach.

In this paper, a novel speech enhancement algorithm based on the MAP estimation is proposed. The proposed speech enhancer adaptively changes the speech spectral density used in the MAP estimation according to the sum of the observed power spectra. In a speech segment, the speech spectral density approaches to Rayleigh distribution to keep the quality of the enhanced speech. While in a non-speech segment, it approaches to an exponential distribution to reduce noise effectively. Furthermore, when the noise is super-Gaussian, we modify the width of Gaussian so that the Gaussian model with the modified width approximates the distribution of the super-Gaussian noise. This technique is effective in suppressing residual noise well. From computer experiments, we confirm the effectiveness of the proposed method.

Digital hearing aid users often complain of difficulty in understanding speech in the presence of background noise. To improve speech perception in a noisy environment, various speech enhancement algorithms have been applied in digital hearing aids. In this study, a speech enhancement algorithm using modified spectral subtraction and companding is proposed for digital hearing aids. We adjusted the biases of the estimated noise spectrum, based on a subtraction factor, to decrease the residual noise. Companding was applied to the channel of the formant frequency based on the speech presence indicator to enhance the formant. Noise suppression was achieved while retaining weak speech components and avoiding the residual noise phenomena. Objective and subjective evaluation under various environmental conditions confirmed the improvement due to the proposed algorithm. We tested segmental SNR and Log Likelihood Ratio (LLR), which have higher correlation with subjective measures. Segmental SNR has the highest and LLR the lowest correlation of the methods tested. In addition, we confirmed by spectrogram that the proposed method significantly reduced the residual noise and enhanced the formants. A mean opinion score that represented the global perception score was tested; this produced the highest quality speech using the proposed method. The results show that the proposed speech enhancement algorithm is beneficial for hearing aid users in noisy environments.

Recently, the decision feedback channel estimation based on the minimum mean square error criterion (DF-MMSE-CE) using a fixed DF filter coefficient has been proposed to improve the channel estimation accuracy for DS-CDMA with frequency-domain equalization (FDE). In this paper, we propose adaptive DF (ADF)-MMSE-CE, in which the DF filter coefficient is adapted to changing channel conditions based on a recursive least square (RLS) algorithm. Furthermore, the channel estimate is phase corrected upon the reception of the periodically inserted pilot chip blocks. The average BER performance of DS-CDMA with MMSE-FDE using ADF-MMSE-CE is evaluated by computer simulation in a frequency-selective Rayleigh fading channel and the simulation results show that our proposed scheme is very robust against fast fading.

In this paper, we propose a new concurrency control protocol for parallel B-tree structures capable reducing the cost of structure-modification-operation (SMO) compared to the conventional protocols such as ARIES/IM and INC-OPT. We call this protocol the MARK-OPT protocol, since it marks the lowest SMO occurrence point during optimistic latch-coupling operations. The marking reduces middle phases for spreading an X latch and removes needless X latches. In addition, we propose three variations of the MARK-OPT, which focus on tree structure changes from other transactions. Moreover, the proposed protocols are deadlock-free and satisfy the physical consistency requirement for B-trees. These indicate that the proposed protocols are suitable as concurrency control protocols for B-tree structures. To compare the performance of the proposed protocols, the INC-OPT, and the ARIES/IM, we implement these protocols on an autonomous disk system adopting the Fat-Btree structure, a form of parallel B-tree structure. Experimental results in various environments indicate that the proposed protocols always improve system throughput, and 2P-REP-MARK-OPT is the most useful protocol in high update environment. Additionally, to mitigate access skew, data should be migrated between PEs. We also demonstrate that MARK-OPT improves the system throughput under the data migration and reduces the time for data migration to balance load distribution.

The specification of a Multi-Agent System (MAS) involves the identification of a large number of entities and their relationships. This is a non-trivial task that requires managing different views of the system. Many problems concerning this issue originate in the presence of contradictory goals and tasks, inconsistencies, and unexpected behaviours. Such troublesome configurations should be detected and prevented during the development process in order to study alternative ways to cope with them. In this paper, we present methods and tools that support the management of contradictions during the analysis and design of MAS. Contradiction management in MAS has to consider both individual (i.e. agent) and social (i.e. organization) aspects, and their dynamics. Such issues have already been considered in social sciences, and more concretely in the Activity Theory, a social framework for the study of interactions in activity systems. Our approach applies knowledge from Activity Theory in MAS, especially its base of contradiction patterns. That requires a formalization of this social theory in order to be applicable in a software engineering context and its adaptation to agent-oriented methodologies. Then, it will be possible to check the occurrence of contradiction patterns in a MAS specification and provide solutions to those situations. This technique has been validated by implementing an assistant for the INGENIAS Development Kit and has been tested with several case studies. This paper shows part of one of these experiments for a web application.

This paper proposes a novel robust fundamental frequency (F0) estimation algorithm based on complex-valued speech analysis for an analytic speech signal. Since analytic signal provides spectra only over positive frequencies, spectra can be accurately estimated in low frequencies. Consequently, it is considered that F0 estimation using the residual signal extracted by complex-valued speech analysis can perform better for F0 estimation than that for the residual signal extracted by conventional real-valued LPC analysis. In this paper, the autocorrelation function weighted by AMDF is adopted for the F0 estimation criterion and four signals; speech signal, analytic speech signal, LPC residual and complex LPC residual, are evaluated for the F0 estimation. Speech signals used in the experiments were an IRS filtered speech corrupted by adding white Gaussian noise or Pink noise whose noise levels are 10, 5, 0, -5 [dB]. The experimental results demonstrate that the proposed algorithm based on complex LPC residual can perform better than other methods in noisy environment.

A speech enhancement method is proposed that can be implemented efficiently due to its use of wavelet packet transform. The proposed method uses a modified spectral subtraction with noise estimation by a least-squares line method and with an overweighting gain per subband with nonlinear structure, where the overweighting gain is used for suppressing the residue of musical noise and the subband is used for applying the weighted values according to the change of signals. The enhanced speech by our method has the following properties: 1) the speech intelligibility can be assured reliably; 2) the musical noise can be reduced efficiently. Various assessments confirmed that the performance of the proposed method was better than that of the compared methods in various noise-level conditions. Especially, the proposed method showed good results even at low SNR.

Probability distribution of the node degree has long been regarded as one of the important metrics to characterize the wireless ad hoc networks. However, there is no exact formulation for this probability distribution when shadow fading effects are considered. In this letter, by deriving the conditional probability of the node degree and the joint distribution of the distances between nodes and a randomly selected reference node (RSRN), we show that the probability distribution of the node degree for the wireless ad hoc networks in the shadow fading environments is binomial.

For a microstrip antenna (MSA) with a ring-shaped slot on formed on the ground plane, downsizing the microstrip patch and expanding the circularly polarized bandwidth have been achieved successfully. The dimensions of the patch are 6.8 mm7.4 mm and the minimum axial ratio (AR) of 0.6 dB is obtained at 6.1 GHz. In addition, its AR is less than 3 dB at the relative bandwidth of 3.5%. The bandwidth of the proposed MSA is twice that of conventional single-feeding circularly polarized MSAs; however, its size is only half that of conventional MSAs.

We survey traffic matrix models, whose elements represent the traffic demand between source-destination pair nodes. Modeling the traffic matrix is useful for multilayer Traffic Engineering (TE) in IP optical networks. Multilayer TE techniques make the network so designed flexible and reliable. This is because it allows reconfiguration of the virtual network topology (VNT), which consists of a set of several lower-layer (optical) paths and is provided to the higher layer, in response to fluctuations (diurnal) in traffic demand. It is, therefore, important to synthetically generate traffic matrices as close to the real ones as possible to maximize the performance of multilayer TE. We compare several models and clarify their applicability to VNT design and control. We find that it is difficult in practice to make an accurate traffic matrix with conventional schemes because of the high cost for data measurement and the complicated calculations involved. To overcome these problems, we newly introduce a simplified traffic matrix model that is practical; it well mirrors real networks. Next, this paper presents our developed server, the IP Optical TE server. It performs multilayer TE in IP optical networks. We evaluate the effectiveness of multilayer TE using our developed IP Optical server and the simplified traffic matrix. We confirm that multilayer TE offers significant CAPEX savings. Similarly, we demonstrate basic traffic control in IP optical networks, and confirm the dynamic control of the network and the feasibility of the IP Optical TE server.

In ad-hoc networks, mobile nodes are limited by a range of radio coverage and have an irregular source of power due to their battery. In ad-hoc networks, there are a lot of situations that all mobile nodes need to agree on their key not at the same time but in part and then merge themselves subsequently. This is because ad-hoc networks have specific features such as mobility and allow various conditions during configuration. In this thesis, we propose MCP (Merging Clusters Protocol), a simple key agreement scheme that can effectively deal with merging different adjacent clusters in mobile ad-hoc networks. When nodes of each cluster have already agreed on their own group keys and intend to merge themselves for further secure communications, MCP can be used in an efficient and secure way. In addition, it can be utilized for efficient group key agreement in a large ad-hoc network. We analyze the security and efficiency of MCP and discuss the experimental results according to practical implementation scenarios.

Several structured peer-to-peer networks have been created to solve the scalability problem of previous peer-to-peer systems such as Gnutella and Napster. These peer-to-peer networks which support distributed hash table functionality construct a sort of structured overlay network, which can cause a topology mismatch between the overlay and the underlying physical network. To solve this mismatch problem, we propose a topology-aware hierarchical overlay framework for DHTs. The hierarchical approach for the overlay is based on the concept that the underlying global Internet is also a hierarchical architecture, that is, a network of networks. This hierarchical approach for the overlay puts forth two benefits: finding data in a physically near place with a high probability, and smaller lookup time. Our hierarchical overlay framework is different from other hierarchical architecture systems in a sense that it provides a specific self-organizing grouping algorithm. Our additional optimization schemes complete the basic algorithm which constructs a hierarchical structure without any central control.

We analytically prove that the error in the channel idle time-based collision probability estimation in face of non-saturated stations is bounded by 2/(CWmin+1) in the IEEE 802.11 wireless LANs (WLANs). This work explicitly quantifies the impact of non-saturation, and the result vindicates the use of the estimation technique in real-life IEEE 802.11 WLANs, in such applications as the acknowledgement-based link adaptation and the throughput optimization through contention window size adaptation.

In this paper, a new subspace-based speech enhancement algorithm is presented. First, we construct a perceptual filterbank from psycho-acoustic model and incorporate it in the subspace-based enhancement approach. This filterbank is created through a five-level wavelet packet decomposition. The masking properties of the human auditory system are then derived based on the perceptual filterbank. Finally, the prior SNR and the masking threshold of each critical band are taken to decide the attenuation factor of the optimal linear estimator. Five different types of in-car noises in TAICAR database were used in our evaluation. The experimental results demonstrated that our approach outperformed conventional subspace and spectral subtraction methods.

The possibility of using three kinds of new type composite materials as material for high speed sliding contacts was investigated. The results of this investigation were compared with the results of the low speed tests that were reported earlier. As a result of the above, it was discovered that for high speed rotation in the range from 0.014 m/s to 2 m/s, the order of merit did not significantly change. Based on this, it was concluded that if solid lubricant is effectively supplied to the sliding surface, the influence by frictional heat generated by high speed is slight. Of the three kinds of composite material, it was clarified that composite material (CMML-1) had the lowest contact resistance and Composite Material (CMML-3) had the lowest maximum frictional coefficient of friction. 'CM' and 'ML' are initialisms for 'Composite Material' and 'Material of Lubrication' respectively. The number that is attached to the material name is a numeric value that was set by this laboratory.

A press-fit connection is a solderless electrical connection technology, which utilizes the mechanical contact force generated between through-holes on a printed circuit board (PCB) and terminals with a width slightly larger than the through-hole diameter. This technology has been widely noted recently as a measure against the "Lead Free Requirement" of materials comprising electric/electronic devices, especially in the area of automobile connector. For the application of this technology to automobile connectors, we have to take into account the severe requirement, such as (1) the adaptation to wider through-hole diameter tolerance range and (2) the establishment of connection reliability for the various PCB surface treatments. As a result, we have determined the minimum and maximum contact forces satisfying the long term connection reliability and designed the terminal shape, which has been refined the N-shape cross section developed before, by using three dimensional finite element methods (FEM). Furthermore, we have developed a new type of hard tin plating on terminals, thus preventing the scraping-off of tin during the insertion process, that could result in a short-circuit on the PCB, for the Organic Solderability Preservative (OSP) treated PCB. The press-fit connector for the automobile airbag Electronic Control Units (ECUs) we developed has been able to transfer to the mass-production phase successfully from August 2005.