In this paper, we propose a novel method based on the second-order conditional maximum a posteriori (CMAP) to improve the performance of the global soft decision in speech enhancement. The conventional global soft decision scheme is found through investigation to have a disadvantage in that the global speech absence probability (GSAP) in that scheme is adjusted by a fixed parameter, which could be a restrictive assumption in the consecutive occurrences of speech frames. To address this problem, we devise a method to incorporate the second-order CMAP in determining the GSAP, which is clearly different from the previous approach in that not only current observation but also the speech activity decisions of the previous two frames are exploited. Performances of the proposed method are evaluated by a number of tests in various environments and show better results than previous work.

This paper describes non-reference objective quality evaluation for noise-reduced speech. First, a subjective test is conducted in accordance with ITU-T Rec. P.835 to obtain the speech quality, the noise quality, and the overall quality of noise-reduced speech. Based on the results, we then propose an overall quality estimation model. The unique point of the proposed model is that the estimation of the overall quality is done only using the previously estimated speech quality and noise quality, in contrast to conventional models, which utilize the acoustical features extracted. Finally, we propose a non-reference objective quality evaluation method using the proposed model. The results of an experiment with different noise reduction algorithms and noise types confirmed that the proposed method gives more accurate estimates of the overall quality compared with the method described in ITU-T Rec. P.563.

This study focuses on system throughput by taking into account the channel interference in IEEE 802.15.3c WPAN, which is based on the hybrid multiple access of CSMA/CA and TDMA, namely CSMA/CA-TDMA. To study the system throughput, we construct a novel analytical model by taking into consideration the channel interference caused by the hidden networks in CSMA/CA-TDMA. The obtained results show that the system throughput achieved by TDMA is highly affected by frame transmission in CSMA/CA. Furthermore, we show that channel interference, which causes a degradation in the system throughput, is a very significant problem in the IEEE 802.15.3c WPAN.

In previous studies, the retrieval accuracy of large image databases has been improved as a result of reducing the semantic gap by combining the input sketch with relevance feedback. A further improvement of retrieval accuracy is expected by combining each stroke, and its order, of the input sketch with the relevance feedback. However, this leaves as a problem the fact that the effect of the relevance feedback substantially depends on the stroke order in the input sketch. Although it is theoretically possible to consider all the possible stroke orders, that would cause a realistic problem of creating an enormous amount of data. Consequently, the technique introduced in this paper intends to improve retrieval efficiency by effectively using the relevance feedback by means of conducting data mining of the sketch considering the similarity in the order of strokes. To ascertain the effectiveness of this technique, a retrieval experiment was conducted using 20,000 images of a collection, the Corel Photo Gallery, and the experiment was able to confirm an improvement in the retrieval efficiency.

This paper presents a pilot-aided channel estimation method which is particularly suitable for mobile WiMAX 802.16e Downlink Partial Usage of Subchannel mode. Based on this mode, several commonly used channel estimation methods are studied and the method of least squares line fitting is proposed. As data of users are distributed onto permuted clusters of subcarriers in the transmitted OFDMA symbol, the proposed channel estimation method utilizes these advantages to provide better performance than conventional approaches while offering remarkably low complexity in practical implementation. Simulation results with different ITU-channels for mobile environments show that depending on situations, enhancement of 5 dB or more in term of SNR can be achieved.

This paper proposes a cross-layer packet scheduling scheme for QoS support in audio-video transmission with IEEE 802.11e HCCA and assesses application-level QoS and QoE of the scheduling scheme under lossy channel conditions. In the proposed scheme, the access point (AP) basically allocates transmission opportunity (TXOP) for each station in a service interval (SI) like the reference scheduler of the IEEE 802.11e standard, which is referred to as the TGe scheme in this paper. In the proposed scheme, however, the AP calculates the number of MAC service data units (MSDUs) arrived in an SI, considering the inter-arrival time of audio samples and that of video frames, which are referred to as media units (MUs), at the application layer. The AP then gives additional TXOP duration in the SI to stations which had audio or video MAC protocol data units (MPDUs) in their source buffers at the end of the previous TXOP. In addition, utilizing video frame information from the application layer, we propose video frame skipping at the MAC-level of a source station. If a station fails to transmit a video MPDU, it drops all the following video MPDUs in the source buffer until the next intra-coded frame comes to the head of the buffer. We compare the reference scheduler (TGe scheme), the proposed packet scheduling scheme with and without the video frame skipping at the source in terms of application-level QoS and QoE. We discuss the effectiveness of the proposed packet scheduling scheme from a viewpoint of QoE as well as QoS. Numerical results reveal that the proposed packet scheduling scheme can achieve higher quality than the TGe scheme under lossy channel conditions. We also show that the proposed scheduling scheme can improve the QoS and QoE by using the video frame skipping at the source. Furthermore, we also examine the effect of SI on the QoS and QoE of the proposed packet scheduling scheme and obtain that the appropriate value of SI is equal to the inter-arrival time of video frame.

In this letter, we develop an analytical model for the drive-thru applications based on the IEEE 802.11p WAVE. The model shows that prioritizing the bitrates via the 802.11e EDCA mechanism leads to significant throughput improvement.

This paper extends the Brezing-Weng method by parameterizing the discriminant D by a polynomial D(x). To date, the maximum of CM discriminant can be adequately addressed is about 14-digits. Thus the degree of the square free part of D(x) has to be sufficiently small. By making the square free part of D(x) a linear monomial, the degree of the square free part is small and by substituting x to some quadratic monomial, pairing-friendly curves with various discriminants can be constructed. In order that a square free part of D(x) is of the form ax, ax has to be a square element as a polynomial representation in a number field. Two methods are introduced to apply this construction. For k = 5, 8, 9, 15, 16, 20, 24 and 28, the proposed method gives smaller ρ value than those in previous studies.

A network of sensors can be used to obtain state based data from the area in which they are deployed. To reduce costs, the data sent via intermediate sensors to a sink are often aggregated. In this letter, we introduce Self-Construction of Aggregation Tree (SCAT) scheme which uses a novel data aggregation scheme utilizing the knowledge of the mobile node and the infrastructure (static node tree) in gathering the data from the mobile node. The static nodes can construct a near- optimal aggregation tree by themselves, using the knowledge of the mobile node, which is a process similar to forming the centralized aggregation tree.

With the emergence of bandwidth-greedy application services, high-speed transport protocols are expected to effectively and aggressively use large amounts of bandwidth in current broadband and multimedia networks. However, when high-speed transport protocols compete with other standard TCP flows, they can occupy most of the available bandwidth leading to disruption of service. To deploy high-speed transport protocols on the Internet, such unfair situations must be improved. In this paper, therefore, we propose a method to improve fairness, called Kyushu-TCP (KTCP), which introduces a non-aggressive period in the congestion avoidance phase to give other standard TCP flows more chances of increasing their transmission rates. This method improves fairness in terms of the throughput by estimating the stably available bandwidth-delay product and adjusting its transmission rate based on this estimation. We show the effectiveness of the proposed method through simulations.

We experimentally studied the polarization mode dispersion (PMD) tolerance of an feed-forward equalizer (FFE) electronic dispersion compensation (EDC) IC in the absence of adaptive control, in 43-Gbit/s RZ-DQPSK transmission. Using a 3-tap FFE IC composed of InP HBTs, differential group delay (DGD) tolerance at a 2-dB Q penalty is shown to be extended from 25 ps to up to 29 ps. When a polarization scrambler is used, the tolerance is further extended to 31 ps. This value is close to the tolerance obtained with adaptive control, without a polarization scrambler.

The multilevel dual-channel (MLDC) not-AND (NAND) flash memories cell structures with asymmetrically-doped channel regions between the source and the drain were proposed to enhance read and program verifying speeds. The channel structure of the MLDC flash memories consisted of two different doping channel regions. The technical computer aided design simulation results showed that the designed MLDC NAND flash cell with asymmetrically-doped channel regions provided the high-speed multilevel reading with a wider current sensing margin and the high-speed program verifying due to the sensing of the discrete current levels. The proposed unique MLDC NAND flash memory device can be used to increase read and program verifying speed.

To improve speech recognition performance, feature transformation based on discriminant analysis has been widely used to reduce the redundant dimensions of acoustic features. Linear discriminant analysis (LDA) and heteroscedastic discriminant analysis (HDA) are often used for this purpose, and a generalization method for LDA and HDA, called power LDA (PLDA), has been proposed. However, these methods may result in an unexpected dimensionality reduction for multimodal data. It is important to preserve the local structure of the data when reducing the dimensionality of multimodal data. In this paper we introduce two methods, locality-preserving HDA and locality-preserving PLDA, to reduce dimensionality of multimodal data appropriately. We also propose an approximate calculation scheme to calculate sub-optimal projections rapidly. Experimental results show that the locality-preserving methods yield better performance than the traditional ones in speech recognition.

With the increasing number of crimes and accidents in which children are becoming involved, there is a growing demand for devices to safeguard children's security by detecting their locations on their way to and from school. This paper proposes a system that uses an IEEE802.15.4-standard network to detect children's locations. To overcome the susceptibility of radio interference from nearby wireless LANs, frequency division multiplexing is applied to this IEEE802.15.4-based network, toward improving data acquisition from terminal units. The effectiveness of the system was field-tested with elementary school students who used about 400 IEEE 802.15.4-compliant terminal units. An experiment verified that the use of frequency division multiplexing in an environment where radio interference by wireless LANs is strong allowed the network to double the success rate of information communication from terminal units relative to that without frequency division multiplexing. In the experiment for detecting elementary schoolers' arrival at and departure from school, the terminal detection rate was 99% and the terminal detection rate on the designated school routes was 90%. These results prove the effectiveness of the system in detecting locations.

Multi-hop Wireless LAN-based mesh network (WMN) provides high capacity and self-configuring capabilities. Due to data forwarding and path selection based on MAC address, WMN requires additional operations to achieve global connectivity using IPv6 address. The neighbor discovery operation over WLAN mesh networks requires repeated all-node broadcasting and this gives rise to a big burden in the entire mesh networks. In this letter, we propose the proxy neighbor discovery scheme for optimized IPv6 communication over WMN to reduce network overhead and communication latency. Using simulation experiments, we show that the control overhead and communication setup latency can be significantly reduced using the proxy-based neighbor discovery mechanism.

We propose a new speech enhancement method suitable for mobile devices used in the presence of various types of noise. In order to achieve high-performance speech recognition and auditory perception in mobile devices, various types of noise have to be removed under the constraints of a space-saving microphone arrangement and few computational resources. The proposed method can reduce both the directional noise and the diffuse noise under the abovementioned constraints for mobile devices by employing a square microphone array and conducting low-computational-cost processing that consists of multiple null beamforming, minimum power channel selection, and Wiener filtering. The effectiveness of the proposed method is experimentally verified in terms of speech recognition accuracy and speech quality when both the directional noise and the diffuse noise are observed simultaneously; this method reduces the number of word errors and improves the log-spectral distances as compared to conventional methods.

A novel turbo coded modulation scheme, called the turbo-APPM, for deep space optical communications is proposed. The proposed turbo-APPM is a serial concatenation of turbo codes, an accumulator and a pulse position modulation (PPM), where turbo codes act as an outer code while the accumulator and the PPM act together as an inner code. The generator polynomial and the puncturing rule for generating turbo codes are chosen to lower the bit error rate. At the receiver, the joint iterative decoding is performed between the inner decoder and the outer turbo decoder. In the outer decoder, local iterative decoding for turbo codes is conducted. Simulation results are presented showing that the proposed turbo-APPM outperforms all previously proposed schemes such as LDPC-APPM, RS-PPM and SCPPM reported in the literature.

Differing from the long-term prediction used in the modern speech codec, the standard of the internet low bit rate codec (iLBC) independently encodes the residual of the linear predictive coding (LPC) frame by frame. In this paper, a complexity scalability design is proposed for the coding of the dynamic codebook search in the iLBC speech codec. In addition, a trade-off between the computational complexity and the speech quality can be achieved by dynamically setting the parameter of the proposed approach. Simulation results show that the computational complexity can be effectively reduced with imperceptible degradation of the speech quality.

This paper presents hybrid type-II automatic repeat request (H-ARQ) for wireless wearable body area networks (BANs) based on ultra wideband (UWB) technology. The proposed model is based on three schemes, namely, high rate optimized rate compatible punctured convolutional codes (HRO-RCPC), Reed Solomon (RS) invertible codes and their concatenation. Forward error correction (FEC) coding is combined with simple cyclic redundancy check (CRC) error detection. The performance is investigated for two channels: CM3 (on-body to on-body) and CM4 (on-body to a gateway) scenarios of the IEEE802.15.6 BAN channel models for BANs. It is shown that the improvement in performance in terms of throughput and error protection robustness is very significant. Thus, the proposed H-ARQ schemes can be employed and optimized to suit medical and non-medical applications. In particular we propose the use of FEC coding for non-medical applications as those require less stringent quality of service (QoS), while the incremental redundancy and ARQ configuration is utilized only for medical applications. Thus, higher QoS is guaranteed for medical application of BANs while allowing coexistence with non-medical applications.

In this letter, we develop an anycast-based emergency service for healthcare wireless sensor networks. The new service could operate with sensors to detect and activate an alarm system based on predefined conditions that are specific to the patient and the disease. The new service is implemented on 802.15.4 ZigBee which usually has large control overhead and long transmission times. To improve the service performance, our service identifies the closest emergency service provider to alleviate the control overhead and achieve immediate assistance when a patient requests for an emergency service. We also increase the reliability of packet transmission by using a Recovery Point scheme. Our simulations and experiment demonstrate that our scheme is efficient and feasible for healthcare wireless sensor networks.