A floorplan is a subdivision of a rectangle into rectangular faces with horizontal and vertical line segments. We call a floorplan room-to-room when adjacencies between rooms are considered. Fujimaki and Takahashi showed that any room-to-room floorplan can be represented as a permutation. In this paper, we give an O(n)-time algorithm that constructs the vertical and the horizontal constraint graphs of a floorplan for a given permutation under this representation.

In our previous study, a distortion measure based variable bit rate (DM-VBR) scheme in waveform interpolation (WI) coders was proposed. In this paper, the repetition method is proposed to estimate non-transmitted parameters instead of the extrapolation method. For the further reduction of slowly evolving waveform (SEW) bit rates, the dimensions of the past parameters, which are different from those of the current parameters, are converted to match the dimension of the current ones. Distortions between interpolated sub-frames and original sub-frames are measured for the reduction of the SEW parameters. And the usefulness of several other distortion measures is also investigated instead of the simple log spectral distortion. Experimental results show that the coder adopting the new schemes offers above 41% bit rate reduction with almost unnoticeable output speech degradation.

This paper presents a color demosaicing method applied to the Bayer pattern color filter array (CFA). Reliable estimation of an edge direction, edge-directed asymmetric interpolation, and the use of color samples at immediate neighbors are considered as the key guidelines for smooth and sharp image restoration. Also, special interest is directed to local areas that are rich in high spatial frequency variations. For suppression of false colors likely to occur in those areas, a hue vector representation is introduced so that the spatial correlation between different color components may be exploited in consistent with the local constant-hue principle. Smoothing is repeated in the hue vector field a few times. Experimental results have shown preferable performances in terms of PSNR, CIELAB color difference, hue angle difference, CIE chromaticity and visual appearance, in particular resulting in less false colors.

We propose the construction of an appearance manifold with embedded view-dependent covariance matrix to recognize 3D objects which are influenced by geometric distortions and quality degradation effects. The appearance manifold is used to capture the pose variability, while the covariance matrix is used to learn the distribution of samples for gaining noise-invariance. However, since the appearance of an object in the captured image is different for every different pose, the covariance matrix value is also different for every pose position. Therefore, it is important to embed view-dependent covariance matrices in the manifold of an object. We propose two models of constructing an appearance manifold with view-dependent covariance matrix, called the View-dependent Covariance matrix by training-Point Interpolation (VCPI) and View-dependent Covariance matrix by Eigenvector Interpolation (VCEI) methods. Here, the embedded view-dependent covariance matrix of the VCPI method is obtained by interpolating every training-points from one pose to other training-points in a consecutive pose. Meanwhile, in the VCEI method, the embedded view-dependent covariance matrix is obtained by interpolating only the eigenvectors and eigenvalues without considering the correspondences of each training image. As it embeds the covariance matrix in manifold, our view-dependent covariance matrix methods are robust to any pose changes and are also noise invariant. Our main goal is to construct a robust and efficient manifold with embedded view-dependent covariance matrix for recognizing objects from images which are influenced with various degradation effects.

Motion-compensated frame interpolation (MCFI) is widely used to smoothly display low frame rate video sequences by synthesizing and inserting new frames between existing frames. The temporal shift interpolation technique (TSIT) is popular for frame interpolation of video sequences that are encoded by a block-based video coding standard such as MPEG-4 or H.264/AVC. TSIT assumes the existence of a motion vector (MV) and may not result in high-quality interpolation for intra-mode blocks that do not have MVs. This paper proposes a new frame interpolation algorithm mainly designed for intra-mode blocks. In order to improve the accuracy of pixel interpolation, the new algorithm proposes sub-pixel interpolation and the reuse of MVs for their refinement. In addition, the new algorithm employs two different interpolation modes for inter-mode blocks and intra-mode blocks, respectively. The use of the two modes reduces ghost artifacts but potentially increases blocking effects between the blocks interpolated by different modes. To reduce blocking effects, the proposed algorithm searches the boundary of an object and interpolates all blocks in the object in the same mode. Simulation results show that the proposed algorithm improves PSNR by an average of 0.71 dB compared with the TSIT with MV refinement and also significantly improves the subjective quality of pictures by reducing ghost artifacts.

This paper proposes a technique for building an effective speech corpus with lower cost by utilizing a statistical multidimensional scaling method. The statistical multidimensional scaling method visualizes multiple HMM acoustic models into two-dimensional space. At first, a small number of voice samples per speaker is collected; speaker adapted acoustic models trained with collected utterances, are mapped into two-dimensional space by utilizing the statistical multidimensional scaling method. Next, speakers located in the periphery of the distribution, in a plotted map are selected; a speech corpus is built by collecting enough voice samples for the selected speakers. In an experiment for building an isolated-word speech corpus, the performance of an acoustic model trained with 200 selected speakers was equivalent to that of an acoustic model trained with 533 non-selected speakers. It means that a cost reduction of more than 62% was achieved. In an experiment for building a continuous word speech corpus, the performance of an acoustic model trained with 500 selected speakers was equivalent to that of an acoustic model trained with 1179 non-selected speakers. It means that a cost reduction of more than 57% was achieved.

In this paper, we propose an image enlargement method by using morphological operators. Our enlargement method is based on the nonlinear frequency extrapolation method (Greenspan et al., 2000) by using a Laplacian pyramid image representation. In this method, the sampling process of input images is modeled as the Laplacian pyramid. A high resolution image is obtained with the finer scale Laplacian that is extrapolated by a nonlinear operation from a low resolution Laplacian. In this paper, we propose a novel nonlinear operation for extrapolation of the finer scale Laplacian. Our nonlinear operation is realized by morphological operators and is capable of generating the finer scale Laplacian, the amplitude of which is proportional to contrasts of edges that appear in the low resolution image. In experiments, the enlargement results given by the proposed method are demonstrated. Compared with the Greenspan's method, the proposed method can recover sharp intensity transients of image edges with small artifacts.

In this paper, we describe a 6-bit 1.6-GS/s flash analog to digital converter (ADC). To reduce the power consumption and active area, we propose a new interpolation architecture using a symmetric three-input comparator. This ADC achieves 5.56 effective bits for input frequencies up to 220 MHz at 1.6 GS/s, and almost five effective bits for 660 MHz input at 1.6 GS/s. Peak INL and DNL are less than 0.5 LSB and 0.45 LSB, respectively. This ADC consumes 85 mW from 1.8 V at 1.6 GS/s and occupies an active area of 0.27 mm2. It is fabricated in 0.18-µm CMOS.

An efficient way to develop large scale speech corpora is to collect phonetically rich ones that have high coverage of phonetic contextual units. The sentence set, usually called as the minimum set, should have small text size in order to reduce the collection cost. It can be selected by a greedy search algorithm from a large mother text corpus. With the inclusion of more and more phonetic contextual effects, the number of different phonetic contextual units increased dramatically, making the search not a trivial issue. In order to improve the search efficiency, we previously proposed a so-called least-to-most-ordered greedy search based on the conventional algorithms. This paper evaluated these algorithms in order to show their different characteristics. The experimental results showed that the least-to-most-ordered methods successfully achieved smaller objective sets at significantly less computation time, when compared with the conventional ones. This algorithm has already been applied to the development a number of speech corpora, including a large scale phonetically rich Chinese speech corpus ATRPTH which played an important role in developing our multi-language translation system.

A sweep spectrum analyzer has been improved over the years, but the fundamental method has not been changed before the 'Super Sweep' method appeared. The 'Super Sweep' method has been expected to break the limitation of the conventional sweep spectrum analyzer, a limit of the maximum sweep rate which is in inverse proportion to the square of the frequency resolution. The superior performance of the 'Super Sweep' method, however, has not been experimentally proved yet. This paper gives the experimental evaluation on the 'Super Sweep' spectrum analyzer, of which theoretical concepts have already been presented by the authors of this paper. Before giving the experimental results, we give complete analysis for a sweep spectrum analyzer and express the principle of the super-sweep operation with a complete set of equations. We developed an experimental system whose components operated in an optimum condition as the spectrum analyzer. Then we investigated its properties, a peak level reduction and broadening of the frequency resolution of the measured spectrum, by changing the sweep rate. We also confirmed that the experimental system satisfactorily detected the spectrum at least 30 times faster than the conventional method and the sweep rate was in proportion to the bandwidth of the base band signal to be analyzed. We proved that the 'Super Sweep' method broke the restriction of the sweep rate put on a conventional sweep spectrum analyzer.

In this review paper we showcase recent activities on silicon photonics science and technology research in Hong Kong regarding two important topical areas--microresonator devices and optical nonlinearities. Our work on silicon microresonator filters, switches and modulators have shown promise for the nascent development of on-chip optoelectronic signal processing systems, while our studies on optical nonlinearities have contributed to basic understanding of silicon-based optically-pumped light sources and helium-implanted detectors. Here, we review our various passive and electro-optic active microresonator devices including (i) cascaded microring resonator cross-connect filters, (ii) NRZ-to-PRZ data format converters using a microring resonator notch filter, (iii) GHz-speed carrier-injection-based microring resonator modulators and 0.5-GHz-speed carrier-injection-based microdisk resonator modulators, and (iv) electrically reconfigurable microring resonator add-drop filters and electro-optic logic switches using interferometric resonance control. On the nonlinear waveguide front, we review the main nonlinear optical effects in silicon, and show that even at fairly modest average powers two-photon absorption and the accompanied free-carrier linear absorption could lead to optical limiting and a dramatic reduction in the effective lengths of nonlinear devices.

In this paper, a CMOS analog-to-digital converter (ADC) with a 6-bit 100 MSPS at 1.8 V is described. The architecture of the proposed ADC is based on a folding type with a resistive interpolation technique for low power consumption. To reduce the power consumption, a folder reduction technique to decrease the number of folding blocks (NFB) by half of the conventional ones, an averaging folder technique, and a compensated resistive interpolation technique are proposed. Further, an auto-switching encoder for efficient digital processing is also presented. With the clock speed of 100 MSPS, the ADC achieves an effective resolution bandwidth (ERBW) of 50 MHz, while consuming only 4.5 mW of power. The measured result of figure-of-merit (FoM) is 0.93 [pJ/convstep]. The active chip occupies an area of 0.28 mm2 in 0.18 µm CMOS technology.

Effect of the post-growth annealing on the morphology of a Ge mesa selectively grown on Si was studied from the viewpoint of near-infrared photodiode applications. By ultrahigh-vacuum chemical vapor deposition, Ge mesas were selectively grown at 600 on Si (001) substrates partially covered with SiO2 masks. The as-grown Ge mesas showed trapezoidal cross-sections having a top (001) surface and {311} sidewall facets, as similar to previous reports. However, after the subsequent post-growth annealing at ~800 in the ultrahigh-vacuum chamber, the mesas were deformed into rounded shapes having a depression at the center and mounds near the edges. Such a deformation cannot be observed for the samples annealed once after cooled and exposed to the air. The residual hydrogen atoms on the Ge surface from the germane (GeH4) decomposition is regarded as a trigger to the observed morphological instability, while the final mesa shape is determined in order to minimize a sum of the surface and/or strain energies.

This paper presents a CMOS A/D converter based on the folding and interpolating technique. A current steering folder composed of differential pairs allows low power operation and an interpolation is used for high speed with low supply voltage. In a folding circuit, only twenty-three MOSFETs are required to have eight reference voltages of an 8-b A/D converter. The interpolation is implemented with a current division technique to generate 32 folding signals. This approach requires much less area and power consumption than other conventional flash A/D converter. The simulation in a 0.35 µm CMOS process achieves 8-b resolution at 250 Msample/s with power consumption 70 mW at 3.3 V power supply. The preliminary experiment indicates the current steering folder and coarse bits operate as expected.

For the purpose of reducing the quantization noise and power consumption of UWB-OFDM transceiver, a new time domain-based interpolator and decimator structure is proposed to realize five-bit D/A and A/D converters in the five-bit 128-tone sigma-delta modulation (SDM) UWB-OFDM transceiver. The five-bit 128-tone SDM UWB-OFDM transceiver using time domain-based interpolator and decimator in place of time spreader and de-spreader can obtain time-domain spread spectrum processing gain and reduce quantization noise simultaneously. The structure of the five-bit 128-tone SDM A/D converter, which employs 32 parallel analog SDM circuits without up-sampling, is designed. Simulation results demonstrate that BER of the proposed five-bit 128-tone SDM D/A and A/D converters based on time domain-based interpolator and decimator scheme can satisfy the performance requirements of the five-bit 128-tone SDM UWB-OFDM transceiver for the QPSK, 16-QAM and 64-QAM modulations.

Cognitive Radios (CR) can recognize the communication environment and switch its communication scheme to more efficiently and flexibly utilize the radio spectrum. The performance of ultra wideband (UWB) degrades if interference is not suppressed properly. We propose here a series of adaptive chirp waveforms in UWB systems. By designing waveform shaping of both linear chirp and non-linear cases, we avoid the estimated spectrum of the on-going applications without the necessity of notch filters, and thus reduce the system complexity. We evaluate system performance of the proposed scheme by simulations and verify that the proposed scheme is a candidate for cognitive UWB systems.

This paper studies a method for transforming ordinary cryptographic primitives to new harder primitives. Such a method is expected to lead to general schemes that make present cryptosystems secure against the attack of quantum computers. We propose a general technique to construct a new function from an ordinary primitive function f with a help of another hard function g so that the resulting function is to be new hard primitives. We call this technique a lifting of f by g. We show that the lifted function is harder than original functions under some simple conditions.

Image authentication is applied to protect the integrity of the digital image. Conventional image authentication mechanisms, however, are unfit for the palette-based color images. Palette-based color images such as GIF images are commonly used for media communications. This article proposes a palette-based color image authentication mechanism. This novel scheme can guarantee the essentials of general authentication schemes to protect palette-based color images. Morphological operations are adopted to draw out the tampered area precisely. According to the experimental results, the images embedded with the authentication data still can preserve high image quality; specifically, the new scheme is highly sensitive to altered areas.

A technique for suppressing the clipping noise of an analogue-to-digital converter (ADC) is proposed to realize a cognitive radio transceiver that offers high sensitivity carrier-sensing. When a large bandwidth cognitive radio transceiver performs carrier-sensing, it must receive a radio wave that includes many primary user transmissions. The radio wave may have high peak-to-average power ratio (PAPR) and clipping noise may be generated. Clipping noise becomes an obstacle to the achievement of high-sensitivity carrier-sensing. In the proposed technique, the original values of the samples clipped by an ADC are estimated by interpolation. Polynomial spline interpolation to the clipped signal is performed in the first step, and then SINC function interpolation is applied to the spline interpolated signal. The performance was evaluated using the signals with various PAPR. It has been found that suppression performance has a dependency on the number of samples clipped at once rather than on PAPR. Although there is an upper limit for the number of samples clipped at once that can be compensated with high accuracy, about 20 dB suppression of clipping noise was achieved with the medium degree of clipping.

For formal verification of large-scale digital circuits, a method using satisfiability checking of logic with equality and uninterpreted functions has been proposed. This logic, however, does not consider specific properties of functions or predicates at all, e.g. associative property of addition. In order to ease this problem, we introduce "equivalence constraint" that is a set of formulas representing the properties of functions and predicates, and check the satisfiability of formulas under the constraint. In this report, we show an algorithm for checking satisfiability with equivalence constraint and also experimental results.