In this paper, authors present new schemes of our proposed Global Open Ethernet (GOE) technology from a viewpoint of improving reliability in metro-area Ethernet environment and show the numerical evidence on their performance results. Although several standardized or vendor proprietary technologies are proposed to improve Ethernet reliability, they still have reliability problems in terms of long failure recovery time (due to forwarding database (FDB) flush and recovery from a root bridge failure on spanning tree protocol), broadcast storm, and packet loss in network reconfiguration. To solve these problems, we introduce three schemes, a Per Destination - Multiple Rapid Spanning Tree Protocol (PD-MRSTP), a GOE Virtual Switch Redundancy Protocol (GVSRP), and an In-Service Reconfiguration (ISR) schemes. PD-MRSTP scheme reduces the failure recovery time by eliminating the need to flush the FDB and to recover from root bridge failures. GVSRP scheme ensures the reliability of connections between a GOE domain and a legacy Ethernet domain. Combined with PD-MRSTP, GVSRP prevents broadcast storm problems due to loops in the inter-domain area. ISR scheme enables in-service bridge replacement and upgrade without packet loss. Evaluating our prototype system, we obtained the following remarkable performance results. The GOE network using PD-MRSTP scheme delivered a fast failure recovery performance (4 ms) independent of the number of MAC address entries, whereas the legacy Ethernet network took 522 ms when a bridge had 6000 MAC address entries. Since we found that the failure recovery time increased in proportion to the number of MAC address entries, the one in large carrier network having one million of MAC address entries would take several tens of seconds. Thus using PD-MRSTP can reduce failure recovery time one ten-thousandth comparing with that of legacy Ethernet. In addition, evaluation of the ISR scheme demonstrated that a network can be upgraded with zero packet loss. Therefore, a GOE-based VPN is a promising alternative to other Ethernet VPNs for its reliability and stability.

In ASR (Automatic Speech Recognition) applications, one of the most important issues in the real-time beamforming of microphone arrays is the inability to capture the whole acoustic dynamics via a finite-length of data and a finite number of array elements. For example, the reflected source signal impinging from the side-lobe direction presents a coherent interference, and the non-minimal phase channel dynamics may require an infinite amount of data in order to achieve perfect equalization (or inversion). All these factors appear as uncertainties or un-modeled dynamics in the receiving signals. Traditional adaptive algorithms such as NLMS that do not consider these errors will result in performance deterioration. In this paper, a time domain beamformer using H∞ filtering approach is proposed to adjust the beamforming parameters. Furthermore, this work also proposes a frequency domain approach called SPFDBB (Soft Penalty Frequency Domain Block Beamformer) using H∞ filtering approach that can reduce computational efforts and provide a purified data to the ASR application. Experimental results show that the adaptive H∞ filtering method is robust to the modeling errors and suppresses much more noise interference than that in the NLMS based method. Consequently, the correct rate of ASR is also enhanced.

Employing noise masking threshold (NMT) to adapt a speech enhancement system has become popular due to the advantage of rendering the residual noise to perceptually white. Most methods employ the NMT to empirically adjust the parameters of a speech enhancement system according to the various properties of noise. In this article, without any predefined empirical factor, an explicit-form gain factor for a frequency bin is derived by perceptually constraining the residual noise below the NMT in spectral domain. This perceptual constraint preserves the spectrum of noisy speech when the level of residual noise is less than the NMT. If the level of residual noise exceeds the NMT, then the spectrum of noisy speech is suppressed to reduce the corrupting noise. Experimental results show that the proposed approach can efficiently remove the added noise in cases of various noise corruptions, and almost free from musical residual noise.

We present a novel approach for single-channel noise reduction of speech signals contaminated by additive noise. In this approach, the system requires speech samples to be uttered in advance by the same speaker as that of the input signal. Speech samples used in this method must have enough phonetic variety to reconstruct the input signal. In the proposed method, which we refer to as referential reconstruction, we have used a small database created from examples of speech, which will be called reference signals. Referential reconstruction uses an example-based approach, in which the objective is to find the candidate speech frame which is the most similar to the clean input frame without noise, although the input frame is contaminated with noise. When candidate frames are found, they become final outputs without any special processing. In order to find the candidate frames, a correlation coefficient is used as a similarity measure. Through automatic speech recognition experiments, the proposed method was shown to be effective, particularly for low-SNR speech signals corrupted with white noise or noise in high-frequency bands. Since the direct implementation of this method requires infeasible computational cost for searching through reference signals, a coarse-to-fine strategy is introduced in this paper.

This paper presents a personal identification method based on BMME and LDA for images acquired at anterior and posterior occlusion expression of teeth. The method consists of teeth region extraction, BMME, and pattern recognition for the images acquired at the anterior and posterior occlusion state of teeth. Two occlusions can provide consistent teeth appearance in images and BMME can reduce matching error in pattern recognition. Using teeth images can be beneficial in recognition because teeth, rigid objects, cannot be deformed at the moment of image acquisition. In the experiments, the algorithm was successful in teeth recognition for personal identification for 20 people, which encouraged our method to be able to contribute to multi-modal authentication systems.

This paper demonstrates a feedforward active substrate noise cancelling technique using a power supply di/dt detector. Since the substrate is usually tied with the ground line with a low impedance, the substrate noise is closely related to the ground bounce which is proportional to the di/dt when inductance is dominant on the ground line impedance. Our active cancelling detects the di/dt of the power supply, and injects an anti-phase current into the substrate so that the di/dt-proportional substrate noise is cancelled out. Our first trial shows that 34% substrate noise reduction is achieved on our test circuit, and the theoretical analysis shows that the optimized canceller design will enhance the substrate noise suppression ratio up to 56%.

Lee et al. recently proposed the first identity-based key agreement protocols for a multiple PKG environment where each PKG has different domain parameters in ICCSA 2005. However, this letter demonstrates that Lee et al.'s scheme does not include the property of implicit key authentication which is the fundamental security requirement, making it vulnerable to an impersonation attack.

This paper describes the author's perspective on multimedia quality prediction methodologies for multimedia communications in advanced mobile and internet protocol (IP)-based telephony, and reports related experiments and trials. First, the paper describes the need for perceptual QoS (Quality of Service) assessment in which various quality factors in multimedia communications for advanced mobile and IP-based telephony are analyzed. Then an objective quality prediction scheme is proposed from the viewpoints of quality measurement tools for each quality factor and an opinion model for compound quality factors in mobile and IP-based communications networks. Finally, the author's current trials of measurement tools and opinion models are described.

Flash bulk files downloading in style of P2P through perpendicular pattern becomes more popular recently. Many peers download different pieces of shared files from the source in parallel. They try to reconstruct complete files by exchanging needed pieces with other downloading peers. The throughput of entire downloading community, as well as the perceived downloading rate of each peer, greatly depends on uploading bandwidth contributed by every individual peer. Unfortunately, without proper built-in incentive mechanism, peers inherently tend to relentlessly download while intentionally limiting their uploading bandwidth. In this paper, we propose a both effective and efficient incentive approach--Reciprocity, which is only based on end-to-end measurement and reaction: a peer caps uploading rate to each of its peers at the rate that is proportional to its downloading rate from that one. It requires no centralized control, or electronic monetary payment, or certification. Preliminary experiments' results reveal that this approach offers favorable performance for cooperative peers, while effectively punishing defective ones.

We consider the problem of computing the optimal swap edges of a shortest-path tree. This problem arises in designing systems that offer point-of-failure shortest-path rerouting service in presence of a single link failure: if the shortest path is not affected by the failed link, then the message will be delivered through that path; otherwise, the system will guarantee that, when the message reaches the node where the failure has occurred, the message will then be re-routed through the shortest detour to its destination. There exist highly efficient serial solutions for the problem, but unfortunately because of the structures they use, there is no known (nor foreseeable) efficient distributed implementation for them. A distributed protocol exists only for finding swap edges, not necessarily optimal ones. We present two simple and efficient distributed algorithms for computing the optimal swap edges of a shortest-path tree. One algorithm uses messages containing a constant amount of information, while the other is tailored for systems that allow long messages. The amount of data transferred by the protocols is the same and depends on the structure of the shortest-path spanning-tree; it is no more, and sometimes significantly less, than the cost of constructing the shortest-path tree.

The performance of a feedback-type adaptive array antenna (AAA) system placed only at a base station (BS) in an FDD/DS-CDMA system remains insufficiently clear. We evaluate the performance of this system by considering the effect of a rake receiver, spacing distance between antennas, the maximum Doppler frequency (fd), and control delay time (Td) on BER performance. In this system, the mobile station (MS) determines optimum weights of antenna elements and sends them back to BS as feedback information. We assume that the optimum weights are not quantized. Thereby, we estimate the performance degradation of 3GPP transmit diversity system, where the feedback information is quantized using a few bits. Computer simulation results show that the rake receiver achieves better BER performance because of the time diversity effect with rake receiver. The AAA with a wide antenna spacing gives high diversity gain for the received signals. For a high value of fd Td, BER performance becomes worse because weighting factors cannot follow the changing speed of channel characteristics. The degradation in performance of a 3GPP system is clarified.

The "Proactive Desk" is a new human-machine interface for the desktop operations of computers. It provides users with tactile sensation in addition to visual sensation. Two linear induction motors underneath the desk generate a two-dimensional force to move objects and control their positions on the desktop using feedback control, and users feel tactile sensation while handling those objects. In this paper, we examined the effects of adding haptic information to simple mouse operation using the Proactive Desk. In our experiment, we used a button-type visual stimulus with and without haptic information. When using haptic conditions, three types of force feedback pattern were displayed: "Edge," "Resistance to motion" and "Attractive force," and each had three force strength conditions: no, half and full. The subject was asked to push buttons twenty times as the buttons were shown one after the other on the desk as quickly as possible. Consequently, the reaction times for pushing the button for all haptic conditions, except for the half-force condition of "Attractive force," were significantly faster than no-force (without haptic information) condition. This result shows that the haptic information was advantageous for easy operation.

Packet loss and delay cause degradation in the quality of real-time, interactive applications such as video conferencing. Forward error correction (FEC) schemes have been proposed to make the applications more resilient to packet loss, because the time required to recover the lost packets is shorter than that required to retransmit the lost packets. On the other hand, the codec generally used in real-time applications like MPEG4 has the feature that the sending bit rate and the packet size of the traffic vary significantly according to the motion of an object in a video. If the traditional FEC coding, which is calculated on the basis of a fixed-size block, is applied to such applications, a waste of bandwidth and a delay variation are caused and the quality is degraded. In this paper, we propose suitable FEC schemes for visual communication systems using variable bit-rate (VBR) codec and evaluate the effectiveness of these schemes using our prototype implementation and experimental network.

In this paper by using an exactly analytic approach the clock jitter in the feedback path of the continuous time Delta Sigma modulators (CT DSM) is modeled as an additive jitter noise, providing a time invariant model for a jittery CT DSM. Then for various DAC waveforms the power spectral density (psd) of the clock jitter at the output of DAC is derived and by using an approximation the in-band power of the clock jitter at the output of the modulator is extracted. The simplicity and generality of the proposed approach are the main advantages of this paper. The MATALB and HSPICE simulation results confirm the validity of the proposed formulas.

We present a quantitative evaluation of speech quality using the multiplexing scheme for the efficient transmission of voice signals in order to reduce the number of the IP packets carrying voice signals (called VoIP packets) transferred. The multiplexing scheme is applicable to a variety of media gateways controlling the bulk of voice streams over IP-based networks, based on VoIP technology. We speculated that the multiplexing scheme would reduce the degradation of speech quality due to packet loss since it also has a similar effect to interleaving the voice signal streams. However, the interleaving effect for maintaining speech quality in the scheme characterized by the feature of IP-based multiplication is not quantitatively clear. Through our end-to-end quality evaluation results of speech, as transmitted via the multiplexing scheme using dedicated hardware, we confirm the advantages of the multiplexing scheme from the perspective of achieving improved speech quality without increasing the processing delay when considering practical packet loss conditions within an IP-based network.

In this paper, we examine the efficiency of tunneling techniques since they will accelerate multicast deployment. Our motivation is that, despite the many proposals focused on tunneling techniques, their impact on multicast efficiency has yet to be assessed sufficiently. First, the structure of multicast delivery trees is examined based on the seminal work of Phillips et al. [26]. We then quantitatively assess the impact of tunneling, such as loads imposed on the tunnel endpoints and redundant traffic. We also formulate a critical size of multicast island, above which the loads are suddenly diminished. Finally, a unique delivery tree model is introduced, which is so simple yet practical, to better understand the performance of the multicast-related protocols. This paper is the first to formulate the impact of tunneling.

We proposed and evaluated a speech packet loss concealment method which predicts lost segments from speech included in packets either before, or both before and after the lost packet. The lost segments are predicted recursively by using linear prediction both in the forward direction from the packet preceding the loss, and in the backward direction from the packet succeeding the lost segment. Predicted samples in each direction are smoothed by averaging using linear weights to obtain the final interpolated signal. The adjacent segments are also smoothed extensively to significantly reduce the speech quality discontinuity between the interpolated signal and the received speech signal. Subjective quality comparisons between the proposed method and the the packet loss concealment algorithm described in the ITU standard G.711 Appendix I showed similar scores up to about 10% packet loss. However, the proposed method showed higher scores above this loss rate, with Mean Opinion Score rating exceeding 2.4, even at an extremely high packet loss rate of 30%. Packet loss concealment of speech degraded with G.729 coding, and babble noise mixed speech showed similar trends, with the proposed method showing higher qualities at high loss rates. We plan to further improve the performance by using adaptive LPC prediction order depending on the estimated pitch, and adaptive LPC bandwidth expansion depending on the consecutive number of repetitive prediction, among many other improvements. We also plan to investigate complexity reduction using gradient LPC coefficient updates, and processing delay reduction using adaptive forward/bidirectional prediction modes depending on the measured packet loss ratio.

This study presents a new current-mirror sense amplifier (CMSA) design for high-speed static random access memory (SRAM) applications. The proposed CMSA can directly sense the current of memory cell and only needs two transistor stages cascaded from VDD to GND for achieving the low-voltage operation. Moreover, the sensing speed of the proposed CMSA is independent of the bit-line capacitances and is only slightly sensitive to the data-line capacitances. Based on the simulation with using the TSMC 0.25-µm 2P4M CMOS process parameter, the proposed CMSA can effectively work at 500 MHz-1 GHz with working voltage as low as 1.5 V. Simulated results show that the proposed CMSA has a much speed improvement compared with the conventional sense amplifiers. Also, the effectiveness of the proposed CMSA is demonstrated with a read-cycle-only memory system to show the good performance for SRAM applications.

A noise reduction technique to reduce wideband and sinusoidal noise in a noisy speech is proposed. In an actual environment, background noise includes not only wideband noise but also sinusoidal noise, such as ventilation fan and engine noise. In this paper, we propose a new noise reduction system which uses two types of adaptive line enhancers (ALE) and a noise estimation filter (NEF). First, the two ALEs are used to estimate speech components. The first ALE is used to reduce sinusoidal noise superposed on speech and wideband noise, while the second ALE is used to reduce wideband noise superposed on speech. However, since the quality of the speech enhanced by two ALEs is not good enough due to the difficulty in estimating unvoiced sound using the two ALEs, the NEF is used to improve on noise reduction capability. The NEF accurately estimates the background noise from the signal occupied by noise components, which is obtained by subtracting the speech enhanced by two ALEs from noisy speech. The enhanced speech is obtained by subtracting the estimated noise from noisy speech. Furthermore, the noise reduction system with feedback path is proposed to improve further the quality of enhanced speech.

The Feedback-Guided Dynamic Loop Scheduling (FGDLS) algorithm [1] is a recent dynamic approach to the scheduling of a parallel loop within a sequential outer loop. Earlier papers have analysed convergence under the assumption that the workload is a positive, continuous, function of a continuous argument (the iteration number). However, this assumption is unrealistic since it is known that the iteration number is a discrete variable. In this paper we extend the proof of convergence of the algorithm to the case where the iteration number is treated as a discrete variable. We are able to establish convergence of the FGDLS algorithm for the case when the workload is monotonically decreasing.