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.

We study fast inversion of the Paillier encryption function. Especially, we focus only on key generation, and do not modify the Paillier encryption function. We propose three key generation algorithms based on the speeding-up techniques for the RSA encryption function. By using our algorithms, the size of the private CRT exponent is half of that of Paillier-CRT. The first algorithm employs the extended Euclidean algorithm. The second algorithm employs factoring algorithms, and can construct the private CRT exponent with low Hamming weight. The third algorithm is a variant of the second one, and has some advantage such as compression of the private CRT exponent and no requirement for factoring algorithms. We also propose the settings of the parameters for these algorithms and analyze the security of the Paillier encryption function by these algorithms against known attacks. Finally, we give experimental results of our algorithms.

RC4 is the stream cipher proposed by Rivest in 1987, which is widely used in a number of commercial products because of its simplicity and substantial security. RC4 exploits shuffle-exchange paradigm, which uses a permutation S. Many attacks have been reported so far. No study, however, has focused on correlations in the Pseudo-Random Generation (PRGA) between two permutations S and S' with some differences, nevertheless such correlations are related to an inherent weakness of shuffle-exchange-type PRGA. In this paper, we investigate the correlations between S and S' with some differences in the initial round. We show that correlations between S and S' remain before "i" is in the position where the nonzero-bit difference exists in the initial round, and that the correlations remain with non negligible probability even after "i" passed by the position. This means that the same correlations between S and S' will be observed after the 255-th round. This reveals an inherent weakness of shuffle-exchange-type PRGA.

This paper presents a CMOS signal detection circuit for 2.5 Gb/s serial data communication system over FR-4 backplane. This overcomes characteristics deviation of full-wave rectifier-based simple power detection circuits due to data pattern and temperature by using an edge detector and a sample-hold circuit.

LCD Automatic Vision Inspection (AVI) systems automatically detect defect features and measure their sizes via camera vision. AVI systems usually report different measurements on the same defect with some variations on position or rotation mainly because we get different images. This is caused by possible variations in the image acquisition process including optical factors, non-uniform illumination, random noise, and so on. For this reason, conventional area based defect measuring method has some problems in terms of robustness and consistency. In this paper, we propose a new defect size measuring method to overcome these problems. We utilize volume information which is completely ignored in the area based conventional defect measuring method. We choose a bell shape as a defect model for experiment. The results show that our proposed method dramatically improves robustness of defect size measurement. Given proper modeling, the proposed volume based measuring method can be applied to various types of defect for better robustness and consistency.

This paper proposes a de-embedding method for on-chip S-parameter measurements at mm-wave frequency. The proposed method uses only two transmission lines with different length. In the proposed method, a parasitic-component model extracted from two transmission lines can be used for de-embedding for other-type DUTs like transistor, capacitor, inductor, etc. The experimental results show that the error in characteristic impedance between the different-length transmission lines is less than 0.7% above 40 GHz. The extracted pad model is also shown.

The CSMA/CA driven MAC protocols withhold packet transmissions from exposed stations when they detect carrier signal above a certain threshold. This is to avoid collisions at other receiving stations. However, this conservative scheme often exposes many stations unnecessarily, and thus minimizes the utilization of the spatial spectral resource. In this paper, we demonstrate that remote estimation of the status at the active receivers is more effective at avoiding collisions in wireless networks than the carrier sensing. We apply a new concept of the interference range, named as n-tolerant interference range, to guarantee reliable communications in the presence of n (n≥ 0) concurrent transmissions from outside the range. We design a distributed interference preventive MAC ( IP-MAC ) using the n-tolerant interference range that enables parallel accesses from the noninterfering stations for an active communication. In IP-MAC, an exposed station goes through an Interference Potentiality Check (IPC) to resolve whether it is potentially interfering or noninterfering to the active communication. During the resolve operation, IPC takes the capture effect at an active receiver into account with interfering signals from a number of possible concurrent transmissions near that receiver. The performance enhancement offered by IP-MAC is studied via simulations in different environments. Results reveal that IP-MAC significantly improves network performance in terms of throughput and delay.

This paper discusses recent trends in the area of low-power, high-performance A/D conversion. We examine survey data collected over the past twelve years to show that the conversion energy of ADCs has halved every two years, while the speed-resolution product has doubled approximately only every four years. A closer inspection on the impact of technology scaling, and developments in ADC design are then presented to explain the observed trends. Finally, we review opportunities in digitally assisted design for the most popular converter architectures.

In this paper, we describe a novel model training method for one-to-many eigenvoice conversion (EVC). One-to-many EVC is a technique for converting a specific source speaker's voice into an arbitrary target speaker's voice. An eigenvoice Gaussian mixture model (EV-GMM) is trained in advance using multiple parallel data sets consisting of utterance-pairs of the source speaker and many pre-stored target speakers. The EV-GMM can be adapted to new target speakers using only a few of their arbitrary utterances by estimating a small number of adaptive parameters. In the adaptation process, several parameters of the EV-GMM to be fixed for different target speakers strongly affect the conversion performance of the adapted model. In order to improve the conversion performance in one-to-many EVC, we propose an adaptive training method of the EV-GMM. In the proposed training method, both the fixed parameters and the adaptive parameters are optimized by maximizing a total likelihood function of the EV-GMMs adapted to individual pre-stored target speakers. We conducted objective and subjective evaluations to demonstrate the effectiveness of the proposed training method. The experimental results show that the proposed adaptive training yields significant quality improvements in the converted speech.

In this paper, a novel orthogonal frequency-division multiplexing (OFDM) modulator/demodulator based on real-valued discrete Hartley transform (DHT) is presented and implemented for the IEEE 802.11a/g wireless local area network (LAN). Instead of the conventional complex-valued fast Fourier transform (FFT) for OFDM systems, the proposed architecture employs two real-valued fast DHT (FHT) kernels and one post processing unit. By taking advantage of the real-valued operation of FHT, this approach reduces the number of multiplications compared with the radix-2 FFT. The proposed DHT-based modulator/demodulator was designed and fabricated in 0.18-µm CMOS technology with a core area of 928935 µm2. The average power consumption is about 20.16 mW at 20 MHz and 1.8 V supply voltage. Measurement results of the integrated circuit illustrate its superior chip area and power consumption.

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.

We propose a simplified model of real-time joint source-channel coding, which can be used to adaptively determine the quality-optimal code rate of forward error correction (FEC) coding. The objective is to obtain the maximum video quality in the receiver, while taking time-varying packet loss into consideration. To this end, we propose a simplified model of the threshold set of the residual video packet loss rate (RVPLR). The RVPLR is the rate of residual loss of video packets after channel decoding. The threshold set is defined as a set of discrete RVPLRs in which the FEC code rate must be changed in order to maintain minimum distortion during increases or decreases of channel packet loss. Because the closed form of the proposed model is very simple and has one scene-dependent model parameter, a video sender can be easily implemented with the model. To train the scene-dependent model parameters in real-time, we propose a test-run method. This method accelerates the test-run while remaining sufficiently accurate for training the scene-dependent model parameters. By using the proposed model and test-run, the video sender can always find the optimal code rate on the fly whenever there is a change in the packet loss status in the channel. An experiment shows that the proposed model and test-run can efficiently determine the near-optimal code rate in joint source-channel coding.

In this letter, a new CCM material, adding Ni powder to a conventional CCM, for X-band applications is designed and analyzed to improve the SE. To obtain the SE of the fabricated CCM accurately, material constants of the CCM of the permittivity and permeability were extracted using transmission/reflection measurements. Using the material constants derived from the measurement, the SE was calculated and the results were verified using a commercial full-wave three-dimensional electromagnetic wave simulator. The SE of the proposed the CCM was improved by approximately 4 dB in the X band compared to that of a conventional CCM. The CCM proposed in this paper can be applied as a shielding material as well as for housing of various communication systems and electrical instruments.

The bit-error-rate (BER) performance predicted by the semi-analytical evolution technique proposed by Li Ping et al. becomes inaccurate for parallel concatenated coded interleave-division multiple-access (PCC-IDMA) systems. To solve this problem, we develop a novel evolution technique of such systems. Numerical results show that the predicted performance agrees well with the simulation results, and that this technique is useful for system optimization.

The design of microstrip bandpass filters using stepped-impedance resonators (SIRs) is examined. The passband center frequency for the WCDMA-FDD (uplink band) Japanese cellular system is 1950 MHz with a 60-MHz bandwidth. The SIR physical characteristic can be designed using a SIR characteristic chart based on second harmonic suppression. In our filter design, passband design charts were obtained through the design procedure. Tchebycheff and maximally flat bandpass filters of any bandwidth and any number of steps can be designed using these passband design charts. In addition, sharp skirt characteristics in the passband can be realized by having two transmission zeros at both adjacent frequency bands by using open-ended quarter-wavelength stubs at input and output ports. A new even-mode harmonics suppression technique is proposed to enable a wide rejection band having a high suppression level. The unloaded quality factor of the resonator used in the proposed filters is greater than 240.

Reflecting the rapid growth of information technology, the configuration of software applications such as word processors and spreadsheets is both sophisticated and complicated. It is often difficult for users to identify relevant functions in the online manual for a target application. In this paper, we propose a method for question answering that finds functions related to the user's request. To enhance our method, we addressed two "mismatch" problems. The first problem is associated with a mismatch in vocabulary, where the same concept is represented by different words in the manual and in the user's question. The second problem is associated with a mismatch in function. Although the user may have a hypothetical function for a purpose in mind, this purpose can sometimes be accomplished by other functions. To resolve these mismatch problems, we extract terms related to software functions from the Web, so that the user's question can be matched to the relevant function with high accuracy. We demonstrate the effectiveness of our method experimentally.

In this letter, we propose distorted scenes enhancement algorithm in order to provide end user perceptual QoE-guaranteed IPTV service. The block edge detection with weight factor and partition-based local color values method can be applied for the degraded video frames which are affected by network transmission errors such as out of order, jitter, and packet loss to improve QoE efficiently. Based on the result of quality metric after using the distorted scenes enhancement algorithm, the distorted scenes have been restored better than others.

Control Area Network (CAN) development began in 1983 and continues today. The forecast for annual world production in 2008 is approximately 65-67 million vehicles with 10-15 CAN nodes per vehicle on average . Although the CAN network is successful in automobile and industry control because the network provides low cost, high reliability, and priority messages, a starvation problem exists in the network because the network is designed to use a fixed priority mechanism. This paper presents a priority inversion scheme, belonging to a dynamic priority mechanism to prevent the starvation problem. The proposed scheme uses one bit to separate all messages into two categories with/without inverted priority. An analysis model is also constructed in this paper. From the model, a message with inverted priority has a higher priority to be processed than messages without inverted priority so its mean waiting time is shorter than the others. Two cases with and without inversion are implemented in our experiments using a probabilistic model checking tool based on an automatic formal verification technique. Numerical results demonstrate that low-priority messages with priority inversion have better expression in the probability in a full queue state than others without inversion. However, our scheme is very simple and efficient and can be easily implemented at the chip level.

Video coding standard H.264/AVC employs transform coding to explore spatial correlation in inter picture prediction residue. This paper presents a block based DC offset to further explore the correlation in spatially neighboring blocks and provides H.264/AVC an enhanced coding efficiency performance. The proposed method applies DC offset to inter picture prediction residue, and encodes the offset compensated residual signal. The DC offset is derived from reconstructed residue in neighboring blocks. No additional bits are required for the DC offset representation. Simulation results report that the proposed method yields an average of 2.67% bit rate reduction for high resolution video over the H.264 baseline profile.

In the last decade, the technique of packet classification has been widely deployed in various network devices, including routers, firewalls and network intrusion detection systems. In this work, we improve the performance of packet classification by using multiple hash tables. The existing hash-based algorithms have superior scalability with respect to the required space; however, their search performance may not be comparable to other algorithms. To improve the search performance, we propose a tuple reordering algorithm to minimize the number of accessed hash tables with the aid of bitmaps. We also use pre-computation to ensure the accuracy of our search procedure. Performance evaluation based on both real and synthetic filter databases shows that our scheme is effective and scalable and the pre-computation cost is moderate.