In this proposed method, CRC coding is only applied to data unit, not to padded zeros that are to fill up the fixed length of data unit. So improved frame error rate and protocol efficiency are possible instead of increment of receiver complexity.

In this paper a new watermarking technique which is combined with joint photographic experts group (JPEG) encoding system is presented. This method operates in the frequency domain by embedding a pseudo-random sequence of real numbers in a selected set of discrete cosine transform (DCT) coefficients. The embedded sequence is extracted without restoring the original image to fit the trend in the digital still camera (DSC) system. The proposed technique represents a major improvement on methods relying on the comparison between the watermarked and original images. Experimental results show that the proposed watermarking method is robust to several common image processing techniques, including JPEG compression, noise, and blurring. We also implement the whole design by synthesizing with TSMC 1P4M 0.35 µm standard cell. The chip size is 3.0643.064 mm2 for 46374 gate counts. The simulation speed can reach 50 MHz. The power dissipation is 69 mW at 3.3 V 50 MHz.

Many techniques have been proposed for segmenting feature point trajectories tracked through a video sequence into independent motions, but objects in the scene are usually assumed to undergo general 3-D motions. As a result, the segmentation accuracy considerably deteriorates in realistic video sequences in which object motions are nearly degenerate. In this paper, we propose a multi-stage unsupervised learning scheme first assuming degenerate motions and then assuming general 3-D motions and show by simulated and real video experiments that the segmentation accuracy significantly improves without compromising the accuracy for general 3-D motions.

Bit-errors in a subband of a wavelet-based video frame during network transmission affect not only lower-level subbands within the same frame but also the subsequent frames. This is because the video frame is wavelet-transformed image with multi-levels and referenced from later frames. In this paper, we propose a new motion estimation scheme for wavelet-based video called Intra-frame Motion Estimation (IME), in which each subband except the LL subband refers to the 1-level-lower subband in the same orientation within the same frame. This scheme protects video quality by confining the effects of the bit-errors of all subbands, except the LL subband, within a frame. We evaluated the performance of our proposed scheme in a simulated wireless network environment. As a result of tests, it was shown that the proposed IME algorithm performs better than MRME, a motion-compensated video coding scheme for wavelet video, in a heavy motion video sequence, while IME outperforms MRME at a high bit-rate in small motion video sequence.

This letter proposes a low-power tournament branch predictor, in which the number of accesses to the branch predictors (local predictor or global predictor) is reduced. Analysis results with Samsung Memory Compiler show that the proposed branch predictor reduces the power consumption by 24-45%, compared to the conventional tournament branch predictor, not requiring any additional storage arrays, not incurring any additional delay and never harming accuracy.

Radars utilizing ultra-wide-band (UWB) pulses are attractive as an environment measurement method for various applications including household robots. Suitable filtering is essential for accurate ranging, which requires an accurate waveform estimation. This paper presents a high-resolution algorithm of estimating target location and scattered waveforms, whose accuracies are interdependent. The technique relies on iterative improvements of estimated waveforms. Description of the algorithm is followed by statistical simulation examples. The performance of the algorithm is contrasted with conventional ones and statistical bounds. Results indicate that our proposed algorithm has a remarkable performance, which is close to the theoretical limit. Next, we clarify the problem of applying HCT to multiple targets. HCT for multiple targets can not be used as an estimated waveform because of interference waves from other targets. We propose an interference suppression algorithm based on a neural network, and show an application example of the algorithm.

In this paper, a coefficient-parameter reduction method is proposed for invertible deinterlacing with variable coefficients. Invertible deinterlacing, which the authors have developed before, can be used as a preprocess of frame-based motion picture codec, such as Motion-JPEG2000 (MJP2), for interlaced videos. When the conventional field-interleaving is used instead, comb-tooth artifacts appear around edges of moving objects. On the other hand, the invertible deinterlacing technique allows us to suppress the comb-tooth artifacts and also to recover an original picture on demand. As previous works, the authors have developed a variable coefficient scheme with a motion detection filter, which realizes adaptability to local characteristics of given pictures. When applying this deinterlacing technique to an image codec, it is required to send coefficient parameters to receivers for original picture recovery. This work proposes a parameter decimation technique and shows that this reduction approach can be achieved without significant loss of comb-tooth suppression capability and improves the quality at high bit-rate decoding.

In story-based communication, where a message is conveyed in story form, it is important to embody the story with expressive materials. However, it is quite difficult for users to create rich multimedia contents using multimedia editing tools. This paper proposes a web-based multimedia environment, SPOC (Stream-oriented Public Opinion Channel), aiming at helping non-skillful people to convert their stories into TV-like programs very easily. The system can produce a digital camera work for graphics and video clips as well as generate an agent animation automatically according to a narration text. Findings in evaluation experiments showed that SPOC is easy-to-use and easy-to-learn for novice users. Given a short instruction, the subjects not only mastered the operations of the software, but also succeeded in creating highly original programs. In subjective evaluation, the subjects answered that they enjoyed using the software without feeling difficulty. These results suggest that this system reduces user's cost in making a program, and encourages communication in a network community.

This paper presents a newly developed noise post-processing (NPP) algorithm and the results of several tests demonstrating its subjective performance. This NPP algorithm is designed to improve the subjective performance of low bit-rate code excited linear prediction (CELP) decoding under background noise conditions. The NPP algorithm is based on a stationary noise generator and improves the subjective quality of noisy signal input. A backward adaptive detector defines noisy input signal frames from decoded LSF, energy, and pitch parameters. The noise generator estimates and produces stationary noise signals using past line spectral frequency (LSF) and energy parameters. The stationary noise generator has a frame erasure concealment (FEC) scheme designed for stationary noise signals and therefore improves the speech decoder's robustness for frame erasure under background noise conditions. The algorithm has been applied to the following CELP decoders: 1) a candidate algorithm of the ITU-T 4-kbit/s speech coding standard and 2) existing ITU-T standards, the G.729 and G.723.1 series. In both cases, NPP improved the subjective performance of the baseline decoders. Improvements of approximately 0.25 CMOS (CCR MOS: comparison category rating mean opinion score) and around 0.2-0.8 DMOS (DCR MOS: degradation category rating mean opinion score) were demonstrated in the results of our subjective tests when applied to the 4-kbit/s decoder and G.729/G.723.1 decoders respectively. Other test results show that NPP improves the subjective performance of a G.729 decoder by around 0.45 in DMOS under both error-free and frame-erasure conditions, and a further improvement of around 0.2 DMOS is achieved by the FEC scheme in the noise generator.

We propose a new data link protocol with an adaptive frame length control scheme for satellite networks. The wireless communication channel in satellite networks is subject to errors that occur with time variance. The frame length of the data link layer is another important factor that affects throughput performance in dynamic channel environments. If the frame length could be chosen adaptively in response to changes in the dynamically varying satellite channel, maximum throughput could be achieved under both noisy and non-noisy error conditions. So, we propose a frame length control scheme that acts adaptively to counter errors that occur with time variance. We model the satellite channel as a two-state Markov block interference (BI) model. The estimation of the channel error status is based on the short-term bit error rate and the duty cycle of noise bursts. Numerical and computer simulation results show that the proposed scheme can achieve high throughput for both dense and diffuse burst noise channels.

We had previously proposed a fuzzy logic-based buffer management scheme (BMS) called fuzzy early detection (FED), which was designed to improve transmission control protocol (TCP) performance over the unspecified bit rate (UBR) service of asynchronous transfer mode (ATM) networks. Since a weakness in FED was discovered later, we present a refined version of it named FED+ here. Maintaining the design architecture and the algorithm of FED, FED+ further adopts a specific per virtual connection accounting algorithm to achieve its design goals. The effects of TCP implementation, TCP maximum segment size, switch buffer size and network propagation delay on FED+ performance are studied through simulation. Its performance is then compared with those of pure early packet discard (EPD), P-random early detection (P-RED) and FED. Our evaluations show that FED+ is superior to the others if the issues of efficiency, fairness, robustness, buffer requirement and the ease of tuning control parameters of a BMS are considered collectively.

This paper proposes a novel method of identifying the design parameters for a practical implementation of the fair queueing discipline, which is capable of class-level delay control. The notion of class weight is introduced at first, and then the session weights are determined. This two-phase approach is favorable in terms of the scalability;that is, the overall complexity is dependent upon the number of classes only. We propose a packet scheduler referred to as the DPS (Delay-centric Processor Sharing) scheme which employs those design parameters to deliver class-wise delay bound services. The associated admission policy for delay guarantee is also derived. System analysis and derivation of the parameters have their origins in the understanding of the so-called system equation, which describes the dynamics of the class-level service share. The proposed design parameters are QoS-aware in that they are consistently refined depending on the system status. Several numerical and simulation results show that the DPS scheme is advantageous over other ones in terms of both resource efficiency and the robustness. Concerning the scalability, we show that an alternative tagging process of the DPS scheme is implementable with O(1) complexity with no significant degradation in delay performance.

Applying the formerly proposed classification framework for first order line search optimization techniques we introduce novel superlinear first order line search methods. Novelty of the methods lies in the line search subproblem. The presented line search subproblem features automatic step length and momentum adjustments at every iteration of the algorithms realizable in a single step calculation. This keeps the computational complexity of the algorithms linear and does not harm the stability and convergence of the methods. The algorithms have none or linear memory requirements and are shown to be convergent and capable of reaching the superlinear convergence rates. They were practically applied to artificial neural network training and compared to the relevant training methods within the same class. The simulation results show satisfactory performance of the introduced algorithms over the standard and previously proposed methods.

This paper proposes a noise robust speech recognition method using prosodic information. In Japanese, the fundamental frequency (F0) contour represents phrase intonation and word accent information. Consequently, it conveys information about prosodic phrases and word boundaries. This paper first describes a noise robust F0 extraction method using the Hough transform, which achieves high extraction rates under various noise environments. Then it proposes a robust speech recognition method using multi-stream HMMs which model both segmental spectral and F0 contour information. Speaker-independent experiments are conducted using connected digits uttered by 11 male speakers in various kinds of noise and SNR conditions. The recognition error rate is reduced in all noise conditions, and the best absolute improvement of digit accuracy is about 4.5%. This improvement is achieved by robust digit boundary detection using the prosodic information.

In wireless communication systems where users compete for limited bandwidth, radio resource control is essential for throughput enhancement and delay reduction. In this paper, we present a game-theoretical approach to distributed resource control in CDMA systems. Incomplete information about channel conditions is considered. The resource control problem is formulated as a noncooperative game of incomplete information, with which the existence and uniqueness of the Bayesian Nash equilibrium (BNE) of the game is investigated. Since the equilibrium is Pareto inefficient, we propose a pricing policy to the resource control game by adding a penalty price to user's transmission cost. With the adoption of the price, user's aggressive behavior is depressed, and Pareto improvement is achieved. Also the Pareto efficient BNE of the game with pricing is studied. Simulation results show that users can obtain higher throughput and lower average packet transmission delay by proper pricing policy. It is also verified that the scheme of pricing policy is robust when information of channel conditions is inaccurate.

We propose Max-Plus Linear (MPL) systems with selective parameters that can describe a certain class of Timed Petri nets (TPN). In this class, selector and joint places are incorporated with Single-Input and Single-Output Timed Event Graph (SISO TEG) subnets. We confirm that the proposed controller effectively works taking into account practical constraints through a numerical example.

Environment measurement is an important issue for various applications including household robots. Pulse radars are promising candidates in a near future. Estimating target shapes using waveform data, which we obtain by scanning an omni-directional antenna, is known as one of ill-posed inverse problems. Parametric methods such as Model-fitting method have problems concerning calculation time and stability. We propose a non-parametric algorithm for high-resolution estimation of target shapes in order to solve the problems of parametric algorithms.

The model for the process of F0 contour generation, first proposed by Fujisaki and his coworkers, has been successfully applied to Standard Chinese, which is a typical tone language with a distinct feature that both positive and negative tone commands are required. However, the inverse problem, viz., automatic derivation of the model parameters from an observed F0 contour of speech, cannot be solved analytically. Moreover, the extraction of model parameters for Standard Chinese is more difficult than for Japanese and English, because the polarity of tone commands cannot be inferred directly from the F0 contour itself. In this paper, an efficient method is proposed to solve the problem by using information on syllable timing and tone labels. With the same framework as for the successive approximation method proposed for Japanese and English, the method presented here for Standard Chinese is focused on the first-order estimation of tone command parameters. A set of intra-syllable and inter-syllable rules are constructed to recognize the tone command patterns within each syllable. The experiment shows that the method works effectively and gives results comparable to those obtained by manual analysis.

In this work, a simple method for extracting MOSFET threshold voltage, effective channel length and channel mobility by using S-parameter measurement is presented. In the new method, the dependence between the channel conductivity and applied gate voltage of the MOSFET device is cleverly utilized to extract the threshold voltage, while biasing the drain node of the device at zero voltage during measurement. Moreover, the effective channel length and channel mobility can also be obtained with the same measurement. Furthermore, all the physical parameters can be extracted directly on the modeling devices without relying on specifically designed test devices. Most important of all, only one S-parameter measurement is required for each device under test (DUT), making the proposed extraction method promising for automatic measurement applications.

In this paper, we propose a dynamic bit-rate reduction scheme for transcoding an MPEG-1 bitstream into an MPEG-4 simple profile bitstream with a typical bit-rate of 384 kbps. For dynamic bit-rate reduction, a significant reduction in the bit-rate is achieved by combining the processes of requantization and frame-skipping. Conventional requantization methods for a homogeneous transcoder cannot be used directly for a heterogeneous transcoder due to the mismatch in the quantization parameters between the MPEG-1 and MPEG-4 syntax and the difference in the compression efficiency between MPEG-1 and MPEG-4. Accordingly, to solve these problems, a new requantization method is proposed for an MPEG-1 to MPEG-4 transcoder consisting of R-Q (rate-quantization) modeling with a simple feedback and an adjustment of the quantization parameters to compensate for the different coding efficiency between MPEG-1 and MPEG-4. For bit-rate reduction by frame-skipping, an efficient method is proposed for estimating the relevant motion vectors from the skipped frames. The conventional FDVS (forward dominant vector selection) method is improved to reflect the effect of the macroblock types in the skipped frames. Simulation results demonstrated that the proposed method combining requantization and frame-skipping can generate a transcoded MPEG-4 bitstream that is much closer to the desired low bit-rate than the conventional method along with a superior objective quality.