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.

A novel concurrent core test approach is proposed to reduce the test cost of SOC. Prior to test, the test sets corresponding to cores under test (CUT) are merged by using the proposed merging algorithm to obtain a minimum merged test set. During test, the proposed scan tree architecture is employed to support the concurrent core test using the merged test set. The approach achieves concurrent core test with one scan input and low hardware overhead. Moreover, the approach does not need any additional test generation, and it can be used in conjunction with general compression/decompression techniques to further reduce test cost. Experimental results for ISCAS 89 benchmarks have proven the efficiency of the proposed approach.

The Ethernet network is widely used and adopted to the access portion or metro area for the reason of new applications for native Ethernet services or its economical advantage. Apart from these applications for native Ethernet, an encapsulation technology to transport legacy services over Ethernet, i.e. TDM over Ethernet, is focused on. In order to apply it to the carrier networks, it is necessary to meet Quality of Service (QoS) requirements, and the consideration of operation, administration and maintenance (OAM) aspects are indispensable. Furthermore, in order for higher reliability, it is required to apply protection function to the networks. We have studied the encapsulation method of TDM signals applied to circuit emulator accommodating TDM signals over Ethernet. In addition, the OAM mechanism and the protection function are studied. This paper shows the frame format, the detail of the OAM mechanism and the protection function, and introduces a developed circuit for adaptation of TDM over Ethernet.

We developed two types of practical maximum-likelihood detectors (MLD) for multiple-input multiple-output (MIMO) systems, using a field programmable gate array (FPGA) device. For implementations, we introduced two simplified metrics called a Manhattan metric and a correlation metric. Using the Manhattan metric, the detector needs no multiplication operations, at the cost of a slight performance degradation within 1 dB. Using the correlation metric, the MIMO-MLD can significantly reduce the complexity in both multiplications and additions without any performance degradation. This paper demonstrates the bit-error-rate performance of these MLD prototypes at a 1 Gbps-order real-time processing speed, through the use of an all-digital baseband 44 MIMO testbed integrated on the same FPGA chip.

This paper discusses propagation delay error, transient response, and power consumption distribution due to inductive effects in optimal buffered on-chip interconnects. Inductive effect is said to be important to consider in deep submicron (DSM) VLSI design. However, study shows that the effect decreases and can be neglected in next technology nodes for such conditions.

In this paper, we introduce a low computing post processing algorithm to simultaneously suppress blocking and ringing artifacts of compressed video sequences. A new regularization function to incorporate smoothness to neighboring pixels is defined, where the function is composed of four sub-functions combined with pixel-based data fidelity and smoothing terms. Therefore, the solution can be obtained without inverse matrix or vector-matrix computation, so that low complexity implementation is possible. In addition, the regularization parameter controlling the relative importance between the data fidelity and the degree of smoothness is estimated from the available overhead information in decoder, such as, macroblock type and quantization step size. The experimental results show the capability and efficiency of the proposed algorithm.

Displaced subdivision surface representation [13] is a new form of representing a polygonal surface model, where a detailed surface model is defined as a scaler-valued displacement map over a smooth domain surface; it puts forth a number of attractive features for editing, geometry compression, animation, scalability, and adaptive rendering of polygonal models. The construction of the smooth domain surface is a challenging task in the conversion process of a detailed polygonal surface model into this representation. In this paper, we propose a new efficient method for defining the smooth domain surface based on -subdivision scheme. The proposed algorithm not only performs better in terms of the quality of the generated surfaces but is computationally more efficient and occupies less memory as compared to the original algorithm [13] and generates surfaces with more levels of detail due to the specific nature of -subdivision when the prescribed target complexity of the generated mesh must not be exceeded. To corroborate the efficiency and the quality of the new technique, the conversion results for several public domain models have been presented.

Lossless video coding is required in the fields of archiving and editing digital cinema or digital broadcasting contents. This paper combines a discrete wavelet transform and adaptive inter/intra-frame prediction in the wavelet transform domain to create multiresolution lossless video coding. Based on the image statistics of the wavelet transform domains in successive frames, inter/intra frame adaptive prediction is applied to the appropriate wavelet transform domain. This adaptation offers superior compression performance. A progressive transmission scheme is also proposed for effective resolution scalability. Experiments on test sequences confirm the effectiveness of the proposed algorithm.

A wireless interface for stacked chips in System-in-a-Package is presented. The interface utilizes inductive coupling between metal inductors. S21 parameters of the inductive coupling are measured between chips stacked in face-up for the first time. Calculations from a theoretical model have good agreement with the measurements. A transceiver circuit for Non-Return-to-Zero signaling is developed to reduce power dissipation. The transceiver is implemented in a test chip fabricated in 0.35 µm CMOS and the chips are stacked in face-up. The chips communicate through the transceiver at 1.2 Gb/s/ch with 46 mW power dissipation at 3.3 V over 300 µm distance. A scaling scenario is derived based on the theoretical model and measurement results. It indicates that, if the communication distance is reduced to 13 µm in 70 nm CMOS, 34 Tbps/mm2 will be obtained.

We propose a cell library development methodology for throughput enhancement of character projection equipment. First, an ILP (Integer Linear Programming)-based cell selection is proposed for the equipment for which both of the CP (Character Projection) and VSB (Variable Shaped Beam) methods are available, in order to minimize the number of electron beam (EB) shots, that is, time to fabricate chips. Secondly, the influence of cell directions on area and delay time of chips is examined. The examination helps to reduce the number of EB shots with a little deterioration of area and delay time because unnecessary directions of cells can be removed. Finally, a case study is shown in which the numbers of EB shots are shown for several cases.

There are many system proposals for satellite-based broadband communications that promise high capacity and ease of access. Many of these proposals require advanced switching technology and signal processing on-board the satellite(s). One solution is based on a geo-synchronous (GEO) satellite system equipped with on-board processing and on-board switching. An important feature of this system is allowing for a maximum number of simultaneous users, hence, requiring effective medium access control (MAC) layer protocols for connection admission control (CAC) and bandwidth on demand (BoD) algorithms. In this paper, an integrated CAC and BoD algorithm is proposed for a broadband satellite communication system with heterogeneous traffic. A detailed modeling and simulation approach is presented for performance evaluation of the integrated CAC and BoD algorithm based on heterogeneous traffic types. The proposed CAC and BoD scheme is shown to be able to efficiently utilize available bandwidth and to gain high throughput, and also to maintain good Grade of Service (GoS) for all the traffic types. The end-to-end delay for real-time traffic in the system falls well within ITU's Quality of Service (QoS) specification for GEO-based satellite systems.

Acoustic modeling in speech recognition uses very little knowledge of the speech production process. At many levels our models continue to model speech as a surface phenomenon. Typically, hidden Markov model (HMM) parameters operate primarily in the acoustic space or in a linear transformation thereof; state-to-state evolution is modeled only crudely, with no explicit relationship between states, such as would be afforded by the use of phonetic features commonly used by linguists to describe speech phenomena, or by the continuity and smoothness of the production parameters governing speech. This survey article attempts to provide an overview of proposals by several researchers for improving acoustic modeling in these regards. Such topics as the controversial Motor Theory of Speech Perception, work by Hogden explicitly using a continuity constraint in a pseudo-articulatory domain, the Kalman filter based Hidden Dynamic Model, and work by many groups showing the benefits of using articulatory features instead of phones as the underlying units of speech, will be covered.

A new system identification based method has been proposed for accurate estimation of vocal tract parameters. An often encountered problem in using the conventional linear prediction analysis is due to the harmonic structure of the excitation source of voiced speech. This harmonic characteristic is coupled with the estimation of autoregressive (AR) coefficients that results in difficulties in estimating the vocal tract filter. This paper models the effective voice source from the residual obtained through the covariance analysis in the first-pass which is then used as input to the second-pass least-square analysis. A better source-filter separation is thus achieved. The formant frequencies and corresponding bandwidths obtained using the proposed method for synthetic vowels are found to be accurate up to a factor of more than three (in percent) compared to the conventional method. Since the source characteristic is taken into account, local variations due to the positioning of analysis window are reduced significantly. The validity of the proposed method is also examined by inspecting the spectra obtained from natural vowel sounds uttered by high-pitched female speaker.

This paper describes a newly developed automatic direction control scheme for bi-directional bus repeaters that uses dynamic collaborative driving techniques. Repeater directions are rapidly determined by detecting the direction of control signal propagation through an additional control signal line that is driven by dynamic collaborative drivers. Application to an on-chip peripheral bus reduces control circuit transistor counts by about 75% and the number of control signal lines by about 50% without loss of speed. Experimental results for a 0.18-µm CMOS implementation indicate that the proposed scheme is four times faster than a conventional scheme with no bi-directional bus repeaters.

This paper presents a shortest path search algorithm using a model of excitable reaction-diffusion dynamics. In our previous work, we have proposed a framework of Digital Reaction-Diffusion System (DRDS)--a model of a discrete-time discrete-space reaction-diffusion system useful for nonlinear signal processing tasks. In this paper, we design a special DRDS, called an "excitable DRDS," which emulates excitable reaction-diffusion dynamics and produces traveling waves. We also demonstrate an application of the excitable DRDS to the shortest path search problem defined on two-dimensional (2-D) space with arbitrary boundary conditions.

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.

Next generation codecs, benchmarked by the H.264/AVC standard, are providing substantial compression efficiency for the coding and transmission of video. Coupled with technologies offering larger transmission bandwidths over DSL, wireless and satellite networks, the capability of delivering high quality video services to the home is now a reality. The perceptual quality of the content delivered over communications networks will be crucial in ensuring a first-class customer experience. It is therefore important to assess the advantages and disadvantages of the optional features offered by next generation codecs. This paper describes a subjective assessment that was carried out to investigate the perceptual effects of switching the in loop de-blocking filter within the H.264/ AVC CODEC on or off. Although the filter is believed to substantially improve the perceptual quality of video, it has been suggested that in some cases negative perceptual effects can be produced. The H.264/AVC architecture allows de-blocking to be switched off in cases where there are limited processing resources or it is considered a negative perceptual effect may be introduced. This paper describes a study that examined the perceptual effects of de-blocking by employing a standardised subjective assessment methodology. The Absolute Category Rating (ACR) method was used to capture Difference Mean Opinion Scores (DMOS) for a range of video. Content was selected to span a wide and representative range of coding complexity. This content was then encoded at a variety of bit-rates to represent high, medium and low qualities. Results were used to examine the end-user perception of video quality when the de-blocking filter is switched on or off. The experimental design allowed the overall effects of the de-blocking filter to be examined and additionally the relationship between content and quality on the filter performance. The experiment found that the performance of the de-blocking filter was content-dependent. Results were used to discuss the advantages and disadvantages of in-loop de-blocking and there is an examination of content properties (e.g. spatial and temporal complexity) that influence the performance of de-blocking.

This paper presents a simple, yet effective hybrid of the minimum mean square error (MMSE) multi-user detection (MUD) and successive interference cancellation (SIC) for direct-sequence code division multiple access (DS-CDMA) systems. The proposed hybrid MUD first divides the users into groups, with each group consisting of users with a close power level. The SIC is then used to distinguish users among different groups, while the MMSE MUD is used to detect signals within each group. To further improve the performance impaired by the propagation errors, an information reuse scheme is also addressed, which can be used in conjunction with the hybrid MMSE/SIC MUD to adequately cancel the multiple access interferences (MAIs) so as to attain more accurate detections. Furthermore, the asymptotic multiuser efficiency (AME), a measure to characterize the near-far resistance capability, is also conducted to provide further insights into the new detectors. Furnished simulations, in both additive white Gaussian noise (AWGN) channels and slow flat Rayleigh fading channels, show that the performances of the proposed hybrid MMSE/SIC detectors, with or without the decision aided scheme, are superior to that of the SIC and, especially, the one with decision aided is close to that of the MMSE MUD but with substantially lower computational complexity.

This paper describes a numerical design procedure of element values of RC polyphase filters with equal minima in stopband and equal ripple in passband. Determination of element values of RC polyphase filters with equal-ripple characteristic have not been solved to the best knowledge of the authors. There found a paper tackling with the problem; however, it can only give sub-optimal solutions via numerical calculation [3]. We propose a numerical element value design procedure for RC polyphase filters with equi-ripple gain in both stopband and passband by using the coefficient matching method. Some design examples are given.

A two-dimensional network for motion detection constructed of simple analog circuits was proposed and designed based on the frog visual system. In the frog visual system, the two-dimensional motion of a moving object can be detected by performing simple information processing in the tectum and thalamus of the frog brain. The measured results of the test chip fabricated by a 1.2 µm complementary metal oxide semiconductor (CMOS) process confirmed the correct operation of the basic circuits in the network. The results obtained with the simulation program with integrated circuit emphasis (SPICE) showed that the proposed network can detect the motion direction and velocity of a moving object. Thus, a chip for two-dimensional motion detection was realized using the proposed network.