Numerous noise suppression methods for speech signals have been developed up to now. In this paper, a new method to suppress noise in speech signals is proposed, which requires a single microphone only and doesn't need any priori-information on both noise spectrum and pitch. It works in the presence of noise with high amplitude and unknown direction of arrival. More specifically, an adaptive noise suppression algorithm applicable to real-life speech recognition is proposed without assuming the Gaussian white noise, which performs effectively even though the noise statistics and the fluctuation form of speech signal are unknown. The effectiveness of the proposed method is confirmed by applying it to real speech signals contaminated by noises.

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Regularization plays a fundamental role in adaptive filtering. There are, very likely, many different ways to regularize an adaptive filter. In this letter, we propose one possible way to do it based on a condition that makes intuitively sense. From this condition, we show how to regularize the recursive least-squares (RLS) algorithm.

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In this paper, we present an approach of detecting speech presence for which the decision rule is based on a combination of multiple features using a sigmoid function. A minimum classification error (MCE) training is used to update the weights adjustment for the combination. The features, consisting of three parameters: the ratio of ZCR, the spectral energy, and spectral entropy, are combined linearly with weights derived from the sub-band domain. First, the Bark-scale wavelet decomposition (BSWD) is used to split the input speech into 24 critical sub-bands. Next, the feature parameters are derived from the selected frequency sub-band to form robust voice feature parameters. In order to discard the seriously corrupted frequency sub-band, a strategy of adaptive frequency sub-band extraction (AFSE) dependant on the sub-band SNR is then applied to only the frequency sub-band used. Finally, these three feature parameters, which only consider the useful sub-band, are combined through a sigmoid type function incorporating optimal weights based on MSE training to detect either a speech present frame or a speech absent frame. Experimental results show that the performance of the proposed algorithm is superior to the standard methods such as G.729B and AMR2.

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Distance to target is fundamental and very important information in numerous engineering fields. Many distance measurement methods using sound use the time delay of a reflected wave, which is measured in reference to the transmitted wave. This method, however, cannot measure short distances because the transmitted wave, which has not attenuated sufficiently by the time the reflected waves are received, suppresses the reflected waves for short distances. Therefore, we proposed an acoustic distance measurement method based on the interference between the transmitted wave and the reflected waves, which can measure distance in a short range. The proposed method requires a cancellation processing for background components due to the spectrum of the transmitted wave and the transfer function of the measurement system in real environments. We refer to this processing as background components cancellation processing (BGCCP). We proposed BGCCP based on subtraction or whitening. However, the proposed method had a limitation with respect to the transmitted wave or additive noise in real environments. In the present paper, we propose an acoustic distance measurement method based on the new BGCCP. In the new BGCCP, we use the calibration of a real measurement system and the whitening processing of the transmitted wave and introduce the concept of the cepstrum to the proposed method in order to achieve robustness. Although the conventional BGCCP requires the recording of the transmitted wave under the condition without targets, the new BGCCP does not have this requirement. Finally, we confirmed the effectiveness of the proposed method through experiments in real environments. As a result, the proposed method was confirmed to be valid and effective, even in noisy environments.

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In this paper, we propose an error resilience scheme for wireless video coding based on adaptive flexible macroblock ordering (FMO) and intra refresh. An FMO explicit map is generated frame-by-frame by using prior information. This information involves estimated locations of guard and burst sections in the channel and estimated effect of error propagation (EEP) from the previous frame to the current frame. In addition, the role of the current frame in propagating an error to the next frame is also considered. A suitable intra refresh rate which is adaptive to the channel state is used to reduce the dependence between frames and thus can stop the EEP. The results in experiments show that the proposed method gains some improvements in terms of peak signal-to-noise rate (PSNR) as compared with some other methods that have not considered the channel condition and the error propagation in generating an FMO map.

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This paper is about a new efficient method for the implementation of convolvers and correlators using the Fermat Number Transform (FNT) and the inverse (IFNT). The latter present advantages compared to Inverse Fast Fourier Transform (IFFT). An efficient state space method for implementing the Inverse FNT (IFNT) over rectangular windows is proposed for the cases where there is a large overlap between the consecutive input signals. This is called Inverse Generalized Sliding Fermat Number Transform (IGSFNT) and is useful for reducing the computational complexity of finite ring convolvers and correlators. This algorithm uses the technique of Generalized Sliding associated to matricial calculation in the Galois Field. The computational complexity of this method is compared with that of standard IFNT.

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In this paper a third-order inter-modulation cancellation technique using Pre-Post-Distortion is proposed to design a wideband high linear low-power LNA in deep submicron. The IM3 cancellation is achieved by post-distorting signal inversely after it is pre- distorted in the input trans-conductance stage during amplification process. The operating frequency range of the LNA is 800MHz-5GHz. The proposed technique increases input-referred third-order intercept point (IIP3) and input 1dB Compression point (P_-1dB) to 12-25dBm and -1.18dBm, respectively. Post layout simulation results show a noise figure (NF) of 4.1-4.5dB, gain of 13.7-13.9dB and S11 lower than -13dB while consumes 8mA from 1.2V supply. The LNA is designed in a 65nm standard CMOS technology. The layout schematic shows that the LNA occupies 0.15×0.11mm² of silicon area.

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The effective ISFs of differential LC oscillators are derived under the assumption that the drain-to-source current is linearly dependent on the gate-to-source voltage for transistors operated in saturation. Moreover, a new interpretation of phase noise is given by examining the real vector diagram of the carrier signal, upon which the noise voltage induced by the impulse noise current is superimposed. The distinct feature of our vector diagram lies in that the noise voltage is always parallel with the horizontal axis. From the Fourier transformations of the derived effective ISFs, the phase noise of differential LCoscillators can be formulated with physical meanings in the frequency domain. The proposed theory can well describe the translation of the noise spectra when the noises from the LC-tank, the switching transistors, and the tail current source are converted into the phase noise. Theoretical predictions from our formulas agree well with the simulation results.

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Timed-Release Encryption (TRE) is a kind of time-dependent encryption, where the time of decryption can be controlled. More precisely, TRE prevents even a legitimate recipient decrypting a ciphertext before a semi-trusted Time Server (TS) sends trapdoor s_T assigned with a release time T of the encryptor's choice. Cathalo et al. (ICICS2005) and Chalkias et al. (ESORICS2007) have already considered encrypting a message intended for multiple recipients with the same release time. One drawback of these schemes is the ciphertext size and computational complexity, which depend on the number of recipients N. Ideally, it is desirable that any factor (ciphertext size, computational complexity of encryption/decryption, and public/secret key size) does not depend on N. In this paper, to achieve TRE with such fully constant costs from the encryptor's/decryptor's point of view, by borrowing the technique of Proxy Re-Encryption (PRE), we propose a cryptosystem in which even if the proxy transformation is applied to a TRE ciphertext, the release time is still effective. By sending a TRE ciphertext to the proxy, an encryptor can foist N-dependent computation costs on the proxy. We call this cryptosystem Timed-Release PRE (TR-PRE). This function can be applied to efficient multicast communication with a release time indication.

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This correspondence contributes to some d-form functions and d-form sequences. A property of d-form functions is obtained firstly. Then we present a way to construct d-form sequences and extended d-form sequences with ideal autocorrelation. Based on our result, many sequences with ideal autocorrelation can be constructed by the corresponding difference-balanced d-form functions.

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In this paper, a new construction method of quaternary sequences of even period 2N having the ideal autocorrelation and balance properties is proposed. These quaternary sequences are constructed by applying the inverse Gray mapping to binary sequences of odd period N with the ideal autocorrelation. Autocorrelation distribution of the proposed quaternary sequences is derived. These sequences can be used to construct quaternary sequence families of even period 2N. Family size and the maximum absolute value of correlation spectrum of the proposed quaternary sequence families are also derived.

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We propose an innovative and practically attainable downlink multi-cell MIMO system with distributed transmit beamforming design. The proposed system is referred to as the MIMO-MAP system which is aimed to mitigate the rank deficiency problem of those MIMO wireless channels that can not support high-order multiplexing gains. In the MIMO-MAP system, each mobile station is allowed to receive several independent data streams from multiple access points at the same time and the same frequency. To do this, a set of noise-subspace-based receive beamformers are employed to suppress the interference among the data streams from different access points. On the other hand, if we consider each receive beamformer as part of its associated wireless channel, we virtually reduce the antenna array at each receive mobile station to a single antenna. With this arrangement, we may have the transmit signal dimension high enough to pre-cancel the inter-stream-interferences at each transmit access point. As a result, the MIMO-MAP channel can be decomposed into a large number of independent subchannels which significantly increase the channel capacity.

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Repeaters equipped with on-board digital baseband processing in a time division duplex (TDD) demand short processing time in order to alleviate inter-symbol interference resulting from having a time delay that is greater than the guard time. To accomplish this, the total system delay of the repeater should be minimized as much as possible without distorting signal quality. Conventionally, the finite impulse response (FIR) type of filter is deployed as a channelization filter, but due to the necessity of large numbers of coefficients to fulfill a prerequisite filter response with a sharp transition band characteristic, an unwanted excessive time delay intrinsically occurs. To make the processing delay as low as possible, this paper proposes a method employing a minimum-phase characterized infinite impulse response (IIR) filter whose magnitude response is almost identical to that of the original FIR filter. Furthermore, in order to linearize the phase response of the designed IIR filter, this paper also introduces an all-pass filter cascaded with the IIR filter for digital down-conversion as well as up-conversion. To achieve further simplicity, this paper introduces polyphase-style IIR filters transformed from conventional single IIR filters that have their own all-pass filters in order to linearize the phase response. The computer simulation results verify that the proposed integrated IIR filter exhibits a relatively short processing delay with a minor deterioration in signal quality-like error vector magnitude (EVM) performance.

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This paper proposes a novel hallucination technique for color face image reconstruction in the RGB, YCbCr, HSV and CIELAB color systems. Our hallucination method depends on multilinear principal component analysis (MPCA) with a linear regression model. In the hallucination framework, many color face images are expressed in color spaces. These images can be naturally described as tensors or multilinear arrays. This novel hallucination technique can perform feature extraction by determining a multilinear projection that captures most of the original tensorial input variation. In our experiments, we used facial images from the FERET database to test our hallucination approach which is demonstrated by extensive experiments with high-quality hallucinated color face images. The experimental results show that a correlation between the color channel and the proposed hallucination method can reduce the complexity in the color face hallucination process.

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A new speed enhancement technique for pulsed laser rangefinders based on Lagrange's theorem in group theory using an undersampling method has been developed. In the undersampling method, frequency conversion for high-resolution ranging and digitizing are conducted by sampling a reference frequency signal at the timings of the reception of pulsed light from the target. In the present work, the rangefinder generates different sampling intervals of the reference frequency signal: different numbers of sampling points within the period of a reference signal, over a wide range. This is accomplished by slightly changing the period of the pulsed light emitted, without changing the synthesizer frequency which generates the period. This technique requires a minimum of additional hardware. In this paper, we describe the detail of the selection of the number of sampling points based on Lagrange's theorem. And we demonstrate a possibility of expanding the sampling interval to the point where an aliasing of the harmonic components of the reference signal occurs by simulations that focus on the calculation of the phase of the fundamental frequency of the reference signal. And we report on the results of rangefinder experiments for a reduction in the number of the sampling points. We have achieved a 10-fold enhancement of speed by selecting 10 sampling points over the results from the previous studies that had 100 sampling points within a period of a reference signal. And we have confirmed that the reduction in sampling points has a very little influence on the linearity, which is an acceptable trade-off for achieving the speed enhancement. This technique, based on Lagrange's theorem in group theory, allows us to control the minimum number of samplings required to calculate distances, so that high-speed data acquisition for coarse measurements and normal-speed data acquisition for fine measurements become selectable. Such a system with high flexibility in measurement modes has been developed.

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The adequate evaluation of sound quality is an important issue for the lossy compression codecs, such as MP3. ITU-R Rec BS. 1387-1 (PEAQ — Perceptual Evaluation of Audio Quality) is the most widely used method to evaluate sound quality objectively. However, PEAQ can only be used for mono signals or two channel stereo signals, because it considers only timbral factors when assessing sound quality. This paper introduces an improved objective quality assessment method that can be used for mono signals and multichannel audio signals that considers both “spatial” and “timbral” factors. The “spatial” factors, which measure perceptual distortions in spatial impression, are important to evaluate the quality of multichannel sounds.

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We consider a stabilization problem of a class of input-delayed nonlinear systems that have not only feedforward, but also some non-feedforward nonlinearity. While there are some existing results that deal with input-delayed non-feedforward nonlinear systems, they often assume a small input delay. It has been often the case that for a large input delay, the results are limited to only feedforward systems. In this letter, combined with the LMI approach in [3] and the reduction method in [5], we show that some feedforward and non-feedforward systems with a large delay in the input can be stabilized via the proposed controller.

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Complex-valued Associative Memory (CAM) is an advanced model of Hopfield Associative Memory. The CAM is based on multi-state neurons and has the high ability of representation. Lee proposed gradient descent learning for the CAM to improve the storage capacity. It is based on only the phases of input signals. In this paper, we propose another type of gradient descent learning based on both the phases and the amplitude. The proposed learning method improves the noise robustness and accelerates the learning speed.

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Recently, Yu, Gong and Tang found new constructions of binary sequences of period 4N with optimal autocorrelation magnitude by different interleaved structure of sequences and sequences which had special correlation property, respectively. In this paper, we derive more results on binary sequences of period 4N which also have optimal autocorrelation.

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In this article, we first review key derivation functions specified in NIST SP 800-108 and one proposed by Krawczyk. Then, we propose parallelizable key derivation functions obtained by modifying or using the existing schemes. We also define two measures of efficiency of key derivation functions, and evaluate their performance in terms of the two measures.

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In this letter, we propose a new construction of quaternary low correlation zone (LCZ) sequence set using binary LCZ sequence sets and an inverse Gray mapping. The new construction method provides optimal quaternary LCZ sequence sets even if the employed binary LCZ sequence set is suboptimal. The optimality is improved at the price of alphabet extension.

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In this letter, the reliabilty of the generalized normal-Laplace (GNL) distribution used for modeling the multiple access interference (MAI) plus noise in time-hopping (TH) binary phase-shift keying (BPSK) ultra-wideband (UWB) systems is evaluated in terms of the probability density function and the BER. The multiple access performance of TH-BPSK UWB systems based on GNL model is analyzed. The average BER performance obtained by using GNL approximation well matches with the exact BER results of TH-BPSK UWB systems. The parameter estimates of GNL distribution based on the moments estimation method is also presented.

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