In this paper, we propose a PLSA-based language model for sports-related live speech. This model is implemented using a unigram rescaling technique that combines a topic model and an n-gram. In the conventional method, unigram rescaling is performed with a topic distribution estimated from a recognized transcription history. This method can improve the performance, but it cannot express topic transition. By incorporating the concept of topic transition, it is expected that the recognition performance will be improved. Thus, the proposed method employs a "Topic HMM" instead of a history to estimate the topic distribution. The Topic HMM is an Ergodic HMM that expresses typical topic distributions as well as topic transition probabilities. Word accuracy results from our experiments confirmed the superiority of the proposed method over a trigram and a PLSA-based conventional method that uses a recognized history.

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.

We have previously proposed a noise-robust speaker verification method using fundamental frequency (F0) extracted using the Hough transform. The method also incorporates an automatic stream-weight and decision threshold estimation technique. It has been confirmed that the proposed method is effective for white noise at various SNR conditions. This paper evaluates the proposed method in more practical in-car and elevator-hall noise conditions. The paper first describes the noise-robust F0 extraction method and details of our robust speaker verification method using multi-stream HMMs for integrating the extracted F0 and cepstral features. Details of the automatic stream-weight and threshold estimation method for multi-stream speaker verification framework are also explained. This method simultaneously optimizes stream-weights and a decision threshold by combining the linear discriminant analysis (LDA) and the Adaboost technique. Experiments were conducted using Japanese connected digit speech contaminated by white, in-car, or elevator-hall noise at various SNRs. Experimental results show that the F0 features improve the verification performance in various noisy environments, and that our stream-weight and threshold optimization method effectively estimates control parameters so that FARs and FRRs are adjusted to achieve equal error rates (EERs) under various noisy conditions.

Orbit estimation of space debris, objects of no inherent value orbiting the earth, is a task that is important for avoiding collisions with spacecraft. The Kamisaibara Spaceguard Center radar system was built in 2004 as the first radar facility in Japan devoted to the observation of space debris. In order to detect the smaller debris, coherent integration is effective in improving SNR (Signal-to-Noise Ratio). However, it is difficult to apply coherent integration to real data because the motions of the targets are unknown. An effective algorithm is proposed for echo detection and orbit estimation of the faint echoes from space debris. The characteristics of the evaluation function are utilized by the algorithm. Experiments show the proposed algorithm improves SNR by 8.32 dB and enables estimation of orbital parameters accurately to allow for re-tracking with a single radar.

We present an adaptive dynamic load balancing scheme for 3D texture based sort-last parallel volume rendering on a PC cluster equipped with GPUs. Our scheme exploits not only task parallelism but also data parallelism during rendering by combining the hierarchical data structures (octree and parallel BSP tree) in order to skip empty regions and distribute proper workloads to rendering nodes. Our scheme can also conduct a valid parallel rendering and image compositing in visibility order by employing a 3D clustering algorithm. To alleviate the imbalance when the transfer function is changed, a load rebalancing is inexpensively supported by exchanging only needed data. A detailed performance analysis is provided and scaling characteristics of our scheme are discussed. These show that our scheme can achieve significant performance gains by increasing parallelism and decreasing synchronizing costs compared to the traditional static distribution schemes.

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.

In this paper, we consider a method to enhance the throughput of HSDPA systems in the mixed traffic scenario. A channel-dependent adaptive delay barrier (DB) function is proposed to maximize throughput of best-effort (BE) traffic while satisfying the delay latency of voice over internet protocol (VoIP) service. Simulations show that the proposed channel-adaptive DB function raises the throughput of BE traffic service by 30% compared to the conventional scheme, without degrading the capacity of VoIP service over HSDPA system.

In the subcarrier and power allocation schemes in Multi-Input Multi-Output and Orthogonal Frequency Division Multiple Access (MIMO-OFDMA) systems, only equal fairness among users has been considered and no scheme for proportional data rate fairness has been considered. In this paper, a subcarrier, bit and power allocation scheme is proposed to maximize the total throughput under the constraints of total power and proportional data rate fairness among users. In the proposed scheme, joint subchannel allocation and adaptive bit loading is firstly performed by using singular value decomposition (SVD) of channel matrix under the constraint of users' data throughput requirements, and then adaptive power loading is applied. Simulation results show that effective performance of the system has been improved as well as each throughput is proportionally distributed among users in MIMO-OFDMA systems.

KASUMI is a blockcipher that forms the heart of the 3GPP confidentiality and integrity algorithms. In this paper, we study the security of the five-round KASUMI type permutations, and derive a highly non-trivial security bound against adversaries with adaptive chosen plaintext and chosen ciphertext attacks. To derive our security bound, we heavily use the tools from graph theory. However the result does not show its super-pseudorandomness, this gives us a strong evidence that the design of KASUMI is sound.

Classification of terrain is one of the most important applications of Polarimetric Synthetic Aperture Radar (POLSAR) image analysis. This paper presents a simple method to classify terrain by the use of the correlation coefficients in the circular polarization basis together with the total power of the scattering matrix in the X-band. The reflection symmetry condition that the co-polarized and the cross-polarized correlations are close to zero for natural distributed scatterers is utilized to extract characteristic parameters of small forests or cluster of trees, and oriented urban building blocks with respect to the direction of the radar illumination. Both of these kinds of scatterers are difficult to identify in high resolution POLSAR images of complex urban areas. The indices employed here are the correlation coefficient, a modified coefficient normalized by the reflection symmetric conditional case, and the total power. It is shown that forest areas and oriented building blocks are easily detected and identified. The terrain classification yielded by these combinations is very accurate as confirmed by photographic ground truth images.

In our application targeted here, four on-glass antenna elements are set in an automobile to improve the reception quality of mobile ISDB-T receiver. With regard to the directional characteristics of each antenna, we propose and implement a joint Pre-FFT adaptive array antenna and Post-FFT space diversity combining (AAA-SDC) scheme for mobile ISDB-T receiver. By applying a joint hardware and software approach, a flexible platform is realized in which several system configuration schemes can be supported; the receiver can be reconfigured on the fly. Simulation results show that the AAA-SDC scheme drastically improves the performance of mobile ISDB-T receiver, especially in the region of large Doppler shift. The experimental results from a field test also confirm that the proposed AAA-SDC scheme successfully achieves an outstanding reception rate up to 100% while moving at the speed of 80 km/h.

The combination of deliberate clipping and an adaptive symbol selection scheme (ASSS) can be used to reduce the peak to average power ratio (PAPR) for Orthogonal Frequency Division Multiplexing (OFDM) signals. The probability density function (pdf) of a sample's amplitude of an adaptively selected OFDM signal without over-sampling has been considered to be approximately equal to the Rayleigh pdf. In this letter, we derive the exact pdf showing the relationship between the probability distribution of the sample's amplitude and the number of candidate OFDM symbols for ASSS. The use of the newly derived pdf can measure the effect of deliberate clipping on the adaptively selected OFDM signal more accurately.

Various wireless systems are being developed to meet users' needs, and the rapid increase in frequency demand that accompanies the increasing popularity of wireless services means that more effective use of frequency resources is urgently needed. However, existing base stations are making no effort to use frequency resources effectively, and cooperation among wireless system base stations is needed to use frequency resources more effectively. Base stations can cooperate more efficiently if they are able to use multiple channels of many wireless systems simultaneously. We propose an autonomous adaptive base station (AABS) that can switch among various wireless systems the way software defined radio (SDR) base stations do. AABS can autonomously select and use the most suitable wireless system on the basis of user traffic and its hardware resources. Moreover, frequency resources are used effectively because AABS prevents unnecessary radio wave transmission when the number of users in the wireless systems decreases. AABS is also suitable for "multi-link communication" because it can use multiple channels of multiple wireless systems simultaneously. We developed AABS prototype and evaluated its performance. Our experimental and computer simulation results show the performance of AABS and its efficiency.

In this paper, we propose an adaptive guard symbol insertion method for one-cell reuse TDMA cellular systems in which co-channel interference is reduced by adaptively selecting the best transmit-pulse waveform with different guard (null-) symbols according to the average error power (AEP) corresponding to signal-to-interference and noise power ratio (SINR), even though the same frequency channel is used at all base stations. Using the proposed system, current TDMA-based systems are readily extensible to one-cell reuse systems, which achieves higher spectrum efficiency. The system capacity is enhanced using the proposed method; moreover, the required qualities such as blocking probability and outage probability are retained.

This paper presents a self-reconfigurable adaptive FIR filter system design using dynamic partial reconfiguration, which has flexibility, power efficiency, advantages of configuration time allowing dynamically inserting or removing adaptive FIR filter modules. This self-reconfigurable adaptive FIR filter is responsible for providing the best solution for realization and autonomous adaptation of FIR filters, and processes the optimal digital signal processing algorithms, which are the low-pass, band-pass and high-pass filter algorithms with various frequencies, for noise removal operations. The proposed stand-alone self-reconfigurable system using Xilinx Virtex4 FPGA and Compact-Flash memory shows the improvement of configuration time and flexibility by using the dynamic partial reconfiguration techniques.

In this research, we focused on fair bandwidth allocation on the Internet. The Internet provides communication services based on exchanged packets. The bandwidth available for each customer is often fluctuated. Fair bandwidth allocation is an important issue for ISPs to gain customer satisfaction. Static bandwidth allocation allows an exclusive bandwidth for specific traffic. Although it gives communications a QoS guarantee, it requires muany bandwidth resources as known as over-provisioning. In contrast with static control, dynamic control allocates bandwidth resources dynamically. It therefore utilizes bandwidth use more effectively. However, it needs control overhead in monitoring traffic and estimating the optimum allocation. The Transmission Control Protocol, or TCP is the dominant protocol on the Internet. It is also equipped with a traffic-rate-control mechanism. An adaptive bandwidth-allocation mechanism must control traffic that is under TCP control. Rapid feedback makes it possible to gain an advantage over TCP control. In this paper, we propose an Adaptive Bandwidth Allocation (ABA) mechanism as a feedback system for MPLS. Our proposal allows traffic to be regulated adaptively as its own weight value which can be assigned by administrators. The feedback bandwidth allocation in the previous work needs round-trip control delay in collecting network status along the communication path. We call this "round-trip feedback control." Our proposal, called "one-way feedback control," collects network status in half the time of roundtrip delay. We compare the performance of our one-way feedback-based mechanism and traditional round-trip feedback control under a simulation environment. We demonstrate the advantages of our rapid feedback control has using experimental results.

Multiple-antenna wireless systems, also known as multiple-input multiple-output (MIMO) cellular networks, can improve the capacity and reliability of communications. To realize these advantages, a packet scheduler should effectively allocate radio resources to users in a fair way. The previously proposed MIMO schedulers have problems such as ignoring traffic arrival process or complexity. We propose a load adaptive multi-output fair queueing (LA-MO-FQ) scheduler, which is based on a fair queueing algorithm with mechanisms for rate selection, compensation of lagging users, and virtual time system. Since some of the scheduler's system parameters are sensitive to the traffic load, it dynamically adjusts them in a way with low complexity so the system performs better. Intensive simulation studies considering the mobility of users and the traffic arrival demonstrate the good performance of LA-MO-FQ. Furthermore, we also propose in this paper some formulae for the time and service fairness comparisons of MIMO schedulers and we use them for comparison with some famous existing schedulers.

We consider power and rate adaptations in multicarrier (MC) direct-sequence code-division multiple-access (DS/CDMA) communications under the assumption that channel state information is provided at both the transmitter and the receiver. We propose, as a power allocation strategy in the frequency domain, to transmit each user's DS waveforms over the user's sub-band with the largest channel gain, rather than transmitting identical DS waveforms over all sub-bands. We then adopt channel inversion power adaptation in the time domain, where the target user's received power level maintains at a fixed value. We also investigate rate adaptation in the time domain, where the data rate is adapted such that a desired transmission quality is maintained. We analyze the BER performance of the proposed power and rate adaptations with fixed average transmission power, and show that power adaptation in both the frequency and the time domains or combined power adaptation in the frequency domain and rate adaptation in the time domain make significant performance improvement over the power adaptation in the frequency domain only. We also compare the performance of the proposed power and rate adaptation schemes in MC-DS/CDMA systems to that of power and rate adapted single carrier DS/CDMA systems with RAKE receiver.

In this letter, we present quasi-Jacket block matrices over GF(2), i.e., binary matrices which all are belong to a class of cocyclic matrices. These matrices are may be useful in digital signal processing, CDMA, and coded modulation. Based on Circular Permutation Matrix (CPM) cocyclic quasi-Jacket block low-density matrix is introduced in this letter which is useful in coding theory. Additionally, we show that the fast algorithm of quasi-Jacket block matrix.

This paper tries to answer the following question: What type of support should be given to an automobile driver when it is determined, via some method to monitor the driver's behavior and the traffic environment, that the driver's intent may not be appropriate to a traffic condition? With a medium fidelity, moving-base driving simulator, three conditions were compared: (a) Warning type support in which an auditory warning is given to the driver to enhance his/her situation recognition, (b) action type support in which an autonomous safety control action is executed to avoid an accident, and (c) the baseline condition in which no driver support is given. Results were as follows: (1) Either type of driver support was effective in accident prevention. (2) Acceptance of driver support functions varied context dependently. (3) Participants accepted a system-initiated automation invocation as long as no automation surprises were possible to occur.