The complexity of the optimal decoding for vector quantization (VQ) in code-division multiple access (CDMA) communications prohibits implementation. It was recently shown in [1] that a suboptimal scheme that combines a soft-output multiuser detector and individual VQ decoders provides a flexible tradeoff between decoder's complexity and performance. The work in [1], however, only considers an AWGN channel model. This paper extends the technique in [1] to a frequency-selective Rayleigh fading channel. Simulation results indicate that such a suboptimal decoder also performs very well over this type of channel.

In orthogonal frequency-division multiplexing (OFDM)-based wireless local area networks (WLANs), phase noise (PHN) and residual frequency offset (RFO) can cause the common phase error (CPE) and the inter-carrier interferences (ICI), which seriously degrade the performance of systems. In this letter, we propose a combined pilot symbol assisted and decision-directed channel estimation scheme based on the least-squares (LS) and the maximum-likelihood (ML) algorithms. Simulation results present that the proposed scheme significantly improves the performance of OFDM-based WLANs.

We propose a new analysis technique against a class of countermeasure using randomized binary signed digit (BSD) representations. We also introduce some invariant properties between BSD representations. The proposed analysis technique can directly recover the secret key from power measurements without information for algorithm because of the invariant properties of BSD representation. Thus the proposed attack is applicable to all countermeasures using BSD representations. Finally, we give the simulation results against some countermeasures using BSD representation such as Ha-Moon method, Ebeid-Hasan method, and the method of Agagliate et al. The results show that the proposed attack is practical analysis method.

In this paper, we will report practical modifications of the side-channel analysis to (EC)DSA [1],[2],[5],[34] that Leadbitter et al. have proposed in [16]. To apply the analyses, we assume that the window method is used in the exponentiation or elliptic curve (EC) scalar multiplication and the side-channel information described in Sect. 3.2 can be collected. So far, the method in [16] hasn't been effective when the size q of a cyclic group used in (EC)DSA is 160 bit long and the window size w < 9. We show that the modified method we propose in this paper is effective even when q is 160 bit long and w=4. This shows that our method is effective for various practical implementations, e.g., that in resource restricted environment like IC card devises. First, we estimate the window size w necessary for the proposed analyses (attacks) to succeed. Then by experiment of the new method, we show that private keys of (EC)DSA can be obtained under the above assumptions, in practical time and with sufficient success rate. The result raises the necessity of countermeasures against the analyses (attacks) in the window method based implementation of (EC)DSA.

To perform a comparative study, we experimented on two differential epitaxial structures, the conventional metamorphic high-electron-mobility-transistor (MHEMT) using the InAlAs/InGaAs/InAlAs structure and the InP-composite-channel MHEMT adopting the InAlAs/InGaAs/InP structure. Compared with the conventional MHEMT, the InP-composite-channel MHEMT shows improved breakdown performance; more than approximately 3.8 V. This increased breakdown voltage can be explained by the lower impact ionization coefficient of the InP-composite-channel MHEMT than that of the conventional MHEMT. The InP-composite-channel MHEMT also shows improved Radio Frequency characteristics of S21 gain of approximately 4.35 dB at 50 GHz, and a cutoff frequency (fT) and a maximum frequency of oscillation (fmax) of approximately 124 GHz and 240 GHz, respectively, were obtained. These are due to decreases in go and gm.

A novel blind channel estimation scheme is proposed for OFDM systems employing PSK modulation. This scheme minimizes the number of possible channels by exploiting the constant modulus property, chooses a best fit over the possible channels by exploiting the finite alphabet property of information signals, and achieves competitive performance with low computational complexity. Results comparing the new scheme with the finite-alphabet based channel estimation are presented.

A time domain (TD) speech enhancement technique to improve SNR in noise-contaminated speech is proposed. Additional supplementary scheme is applied to estimate the degree of noise of noisy speech. This is estimated from a function, which is previously prepared as the function of the parameter of the degree of noise. The function is obtained by least square (LS) method using the given degree of noise and the estimated parameter of the degree of noise. This parameter is obtained from the autocorrelation function (ACF) on frame-by-frame basis. This estimator almost accurately estimates the degree of noise and it is useful to reduce noise. The proposed method is based on two-stage processing. In the first stage, subtraction in time domain (STD), which is equivalent to ordinary spectral subtraction (SS), is carried out. In the result, the noise is reduced to a certain level. Further reduction of noise and by-product noise residual is carried out in the second stage, where blind source separation (BSS) technique is applied in time domain. Because the method is a single-channel speech enhancement, the other signal is generated by taking the noise characteristics into consideration in order to apply BSS. The generated signal plays a very important role in BSS. This paper presents an adaptive algorithm for separating sources in convolutive mixtures modeled by finite impulse response (FIR) filters. The coefficients of the FIR filter are estimated from the decorrelation of two mixtures. Here we are recovering only one signal of interest, in particular the voice of primary speaker free from interfering noises. In the experiment, the different levels of noise are added to the clean speech signal and the improvement of SNR at each stage is investigated. The noise types considered initially in this study consist of the synthesized white and color noise with SNR set from 0 to 30 dB. The proposed method is also tested with other real-world noises. The results show that the satisfactory SNR improvement is attained in the two-stage processing.

This paper presents a test input data compression technique, Selective Low-Care Coding (SLC), which can be used to significantly reduce input test data volume as well as the external test channel requirement for multiscan-based designs. In the proposed SLC scheme, we explored the linear dependencies of the internal scan chains, and instead of encoding all the specified bits in test cubes, only a smaller amount of specified bits are selected for encoding, thus greater compression can be expected. Experiments on the larger benchmark circuits show drastic reduction in test data volume with corresponding savings on test application time can be indeed achieved even for the well-compacted test set.

This paper shows that the Surrounding Gate Transistor (SGT) can be scaled down to decananometer gate lengths by using an intrinsically-doped body and gate work function engineering. Strong gate controllability is an essential characteristics of the SGT. However, by using an intrinsically-doped body, the SGT can realize a higher carrier mobility and stronger gate controllability of the silicon body. Then, in order to adjust the threshold voltage, it is necessary to adopt gate work function engineering in which a metal or metal silicide gate is used. Using a three-dimensional (3D) device simulator, we analyze the short-channel effects and current characteristics of the SGT. We compare the device characteristics of the SGT to those of the Tri-gate transistor and Double-Gate (DG) MOSFET. When the silicon pillar diameter (or silicon body thickness) is 10 nm, the gate length is 20 nm, and the oxide thickness is 1 nm, the SGT shows a subthreshold swing of 63 mV/dec and a DIBL of -17 mV, whereas the Tri-gate transistor and the DG MOSFET show a subthreshold swing of 71 mV/dec and 77 mV/dec, respectively, and a DIBL of -47 mV and -75 mV, respectively. By adjusting the value of the gate work function, we define the off current at VG = 0 V and VD = 1 V. When the off current is set at 1 pA/µm, the SGT can realize a high on current of 1020 µA/µm at VG = 1 V and VD = 1 V. Moreover, the on current of the SGT is 21% larger than that of the Tri-gate transistor and 52% larger than that of the DG MOSFET. Therefore, the SGT can be scaled reliably toward the decananometer gate length for high-speed and low-power ULSI.

A simultaneous equations method is one of active noise control algorithms without estimating an error path. This algorithm requires identification of a transfer function from a reference microphone to an error microphone containing the effect of a noise control filter. It is achieved by system identification of an auxiliary filter. However, the introduction of the auxiliary filter requires more number of samples to obtain the noise control filter and brings a requirement of some undesirable assumption in the multiple channel case. In this paper, a new simultaneous equations method without the identification of the auxiliary filter is proposed. By storing a small number of input signals and error signals, we avoid this identification. Therefore, we can reduce the number of samples to obtain the noise control filters and can avoid the undesirable assumption. From simulation examples, it is verified that the merits of the ordinary method is also retained in the proposed method.

IEEE 802.11 DCF is a contention-based channel access protocol, and medium access delay greatly increases as the number of contending stations in a service area increases. This severely affects on delay-sensitive video applications. In this paper, we focus on MAC-layer solutions for realizing high quality video transmission in 802.11 DCF networks. A new channel access protocol based on the timestamp (TS) of video packets is proposed for real-time video. The TS information is carried by RTP header from the video application to 802.11 MAC. For video packets with the same RTP TS, they are simultaneously transmitted in a single channel access. Additional contention and back-off processes can be avoided because the whole packets of a video frame are completely delivered. The proposed TS-based access protocol can be easily implemented by the DCF with packet bursting. In addition, it is backward compatible to the legacy DCF. Extensive simulations show that the TS-based channel access achieves lower cumulative distributions of application-level video frame delay when compared to the DCF protocol.

This paper investigates limitations of adjacent channel power ratio (ACPR) improvement in predistortion (pre-D) linearizer used with nonlinear RF power amplifiers (PAs) when the PA model is not perfectly acquired in pre-D design. The error between the physical PA and the nonlinear model is expanded by pre-D function and its power spectral density (PSD) works as limitations in ACPR improvement of the pre-D linearizer. An analytical estimation of ACPR limitations in RF PAs driven by digitally modulated input signal is derived using a formulation of autocorrelation function. The analysis technique is validated with the example of the memory polynomial PA model with the quasi-memoryless pre-D linearizer. The technique is also verified by comparing predicted ACPR limitation with measured limitation for a RF PA with 802.11g input signal.

A novel frequency domain training sequence and the corresponding carrier frequency offset (CFO) estimator are proposed for orthogonal frequency division multiplexing (OFDM) systems over frequency-selective fading channels. The proposed frequency domain training sequence comprises two types of pilot tones, namely distinctively spaced pilot tones with high energies and uniformly spaced ones with low energies. Based on the distinctively spaced pilot tones, integer CFO estimation is accomplished. After the subcarriers occupied by the distinctively spaced pilot tones and their adjacent subcarriers are nulled for the sake of interference cancellation, fractional CFO estimation is executed according to the uniformly spaced pilot tones. By exploiting a predefined lookup table making the best of the structure of the distinctively spaced pilot tones, computational complexity of the proposed CFO estimator can be decreased considerably. With the aid of the uniformly spaced pilot tones generated from Chu sequence with cyclically orthogonal property, the ability of the proposed estimator to combat multipath effect is enhanced to a great extent. Simulation results illustrate the good performance of the proposed CFO estimator.

We have proposed two space division multiple access (SDMA) approaches for OFDM signals: "Virtual Subcarrier Assignment (VISA)" and "Preamble Subcarrier Assignment (PASA)," both of which can enhance the system capacity without significant change of transmitter/receiver structures for already-existing OFDM-based standards such as IEEE802.11a. In order to investigate the performance of the proposed approaches in real wireless scenarios, we conducted a measurement campaign to obtain real channel state data at 5-GHz band in an indoor environment. Using the measured channel data, we can make the performance evaluation realistic. In this paper, after the brief overview of the two proposed SDMA approaches, we describe our measurement campaign in detail. Furthermore, we evaluate the performance of VISA-based system and PASA-based system by computer simulations using the measured channel state data and present a comparative study on the performance of the two proposed SDMA approaches in the realistic wireless environment.

This paper describes a novel pilot symbol aided up-link channel estimation scheme for a multi-user MIMO-OFDM system. A novel pilot-symbol pattern is proposed in order to overcome the interference from the multiple antennas of each user. Based on these periodically inserted pilot symbols, the channel state information (CSI) for each entire OFDM data sequence is reconstructed by using the maximum a-posteriori probability (MAP) estimation algorithm. The MAP estimation algorithm exploits channel correlations in time, frequency and space domains, which are obtained from a frequency-selective and time-variant Rayleigh fading channel model with multiple clusters and a defined complex direction of arrival (DOA). Simulation results demonstrate that it achieves almost the same performance as the ideal case by using the MAP-based estimation scheme with the well designed pilot-symbol pattern. Moreover, this model-based estimation scheme is also robust to errors in the estimation of its parameters. It will become one of the strong candidates for use in next generation mobile communication systems.

This paper describes a BIST circuit for testing SoC integrated multi-channel serializer/deserializer (SerDes) macros. A newly developed packet-based PRBS generator enables the BIST to perform at-speed testing of asynchronous data transfers. In addition, a new technique for chained alignment checks between adjacent channels helps achieve a channel-count-independent architecture for verification of multi-channel alignment between SerDes macros. Fabricated in a 0.13-µm CMOS process and operating at > 500 MHz, the BIST has successfully verified all SerDes functions in at-speed testing of 5-Gbps20-ch SerDes macros.

The application of Orthogonal Space-Time Block Codes (O-STBC) as the encoding scheme in the presence of "non-quasi-static" fading was considered. A simple and efficient adaptive method of channel estimation based on the interpolation of estimates acquired at the pre-amble and post-amble of framed blocks of information is developed. Moreover, the proposed method is proven, both theoretically and by simulations, to outperform the alternative of channel tracking, despite its significant low complexity.

Recently, a novel detector was proposed by the authors for code acquisition in non-Gaussian impulsive channels [3], which dramatically outperforms the conventional squared-sum detector; however, it requires exact knowledge of the non-Gaussian noise dispersion. In this paper, a robust detector is proposed, which employs the signs and ranks of the received signal samples, instead of their actual values, and so does not require knowledge of the non-Gaussian noise dispersion. The acquisition performance of the proposed detector is compared with that of the detector of [3] in terms of the mean acquisition time. The simulation results show that the proposed scheme is not only robust to deviations from the true value of the non-Gaussian noise dispersion, but also has comparable performance to that of the scheme of [3] using exact knowledge of the non-Gaussian noise dispersion.

In this letter, we propose a partial minimum mean-squared error (MMSE) with successive interference cancellation (PMMSESIC) method in frequency domain to mitigate ICI caused by channel variation. Each detection, the proposed method detects the symbol with the largest received signal-to-interference-plus-noise ratio (SINR) among all the undetected symbols, using an MMSE detector that considers only the interference of several neithborhood subcarriers. Analysis and simulations show that it outperforms the MMSE method at relatively high Eb/N0 and its performance is close to the MMSE with successive detection (MMSESD) method in relatively low Doppler frequency region.

In this paper, we propose and analyze a scheme for wireless channels, which is the combination of an automatic repeat request (ARQ) scheme and power ramping. The power ramping is used for more reliable downlink data transmission. This technique gradually increases the transmission power at each retransmission attempt. Simulation results demonstrate that when the power step size is 0.5 dB, the average throughput gain may be as high as 2.3% to 5.4% with properly selected parameters.