931 resultados para Frequency domain


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The charactesistics of two-dimension spectra obtained by inductively coupled plasma atomic emission spectrometry (ICP-AES) with charge injection detection (CID) in frequency domain were studied in the present paper. The measurement spectra were Fourier transformed and the frequency distribution of the spectra was obtained. Results showed that the spectra in frequency domain could he divided into two parts:high frequency and low frequency signals. The later stood for measurement spectra and the former for background and noises. However, the high frequecny signals could not be smoothed simply to reduce noises because the background was deteriorated even though the spectral signal did not change significantly.

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We propose a frequency domain adaptive algorithm for
wave separation in wind instruments. Forward and backward travelling waves are obtained from the signals acquired by two microphones placed along the tube, while the
separation ?lter is adapted from the information given by a
third microphone. Working in the frequency domain has a
series of advantages, among which are the ease of design of
the propagation ?lter and its differentiation with respect to
its parameters.
Although the adaptive algorithm was developed as a ?rst
step for the estimation of playing parameters in wind instruments it can also be used, without any modi?cations, for
other applications such as in-air direction of arrival (DOA)
estimation. Preliminary results on these applications will
also be presented.

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In this paper we consider the uplink transmission within CP-assisted (Cyclic Pre¯x) DS-CDMA (Direct Sequence Code Division Multiple Access) systems and we present a frequency-domain MUD (MultiUser Detection) receiver with iterative estimation and compensation of residual frequency errors. The proposed receiver is suitable for broadband wireless systems, with performances that can be close to the single-user MFB (Matched Filter Bound), even for fully loaded systems and/or in the presence of strong interfering signals. The receiver is powerful enough for typical asynchronous scenarios, requiring only a coarse synchronization between users.

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A DS-CDMA (Direct Sequence-Coded Division Multiple Access) system has maximum spectral efficiency if the system is fully loaded (i.e., the number of users is equal to the spreading factor) and we employ signals with bandwidth equal to the chip rate. However, due to implementation constraints we need to employ signals with higher bandwidth, decreasing the system’s spectral efficiency. In this paper we consider prefixassisted DS-CDMA systems with bandwidth that can be significantly above the chip rate. To allow high spectral efficiency we consider highly overloaded systems where the number of users can be twice the spreading factor or even more. To cope with the strong interference levels we present an iterative frequencydomain receiver that takes full advantage of the total bandwidth of the transmitted signals. Our performance results show that the proposed receiver can have excellent performance, even for highly overloaded systems. Moreover, the overall system performance can be close to the maximum theoretical spectral efficiency, even with transmitted signals that have bandwidth significantly above the chip rate.

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On étudie l’application des algorithmes de décomposition matricielles tel que la Factorisation Matricielle Non-négative (FMN), aux représentations fréquentielles de signaux audio musicaux. Ces algorithmes, dirigés par une fonction d’erreur de reconstruction, apprennent un ensemble de fonctions de base et un ensemble de coef- ficients correspondants qui approximent le signal d’entrée. On compare l’utilisation de trois fonctions d’erreur de reconstruction quand la FMN est appliquée à des gammes monophoniques et harmonisées: moindre carré, divergence Kullback-Leibler, et une mesure de divergence dépendente de la phase, introduite récemment. Des nouvelles méthodes pour interpréter les décompositions résultantes sont présentées et sont comparées aux méthodes utilisées précédemment qui nécessitent des connaissances du domaine acoustique. Finalement, on analyse la capacité de généralisation des fonctions de bases apprises par rapport à trois paramètres musicaux: l’amplitude, la durée et le type d’instrument. Pour ce faire, on introduit deux algorithmes d’étiquetage des fonctions de bases qui performent mieux que l’approche précédente dans la majorité de nos tests, la tâche d’instrument avec audio monophonique étant la seule exception importante.

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A MATLAB GUI is presented which is used to help students learn to design controllers in the frequency domain. It complements the author’s two previous GUIs for plotting and identification of systems in the frequency domain. It also incorporates the concept used in the “electronic calculator that makes students think” to assist learning. Positive student feedback affirms that the GUI has helped their understanding.

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Single-carrier (SC) block transmission with frequency-domain equalisation (FDE) offers a viable transmission technology for combating the adverse effects of long dispersive channels encountered in high-rate broadband wireless communication systems. However, for high bandwidthefficiency and high power-efficiency systems, the channel can generally be modelled by the Hammerstein system that includes the nonlinear distortion effects of the high power amplifier (HPA) at transmitter. For such nonlinear Hammerstein channels, the standard SC-FDE scheme no longer works. This paper advocates a complex-valued (CV) B-spline neural network based nonlinear SC-FDE scheme for Hammerstein channels. Specifically, We model the nonlinear HPA, which represents the CV static nonlinearity of the Hammerstein channel, by a CV B-spline neural network, and we develop two efficient alternating least squares schemes for estimating the parameters of the Hammerstein channel, including both the channel impulse response coefficients and the parameters of the CV B-spline model. We also use another CV B-spline neural network to model the inversion of the nonlinear HPA, and the parameters of this inverting B-spline model can easily be estimated using the standard least squares algorithm based on the pseudo training data obtained as a natural byproduct of the Hammerstein channel identification. Equalisation of the SC Hammerstein channel can then be accomplished by the usual one-tap linear equalisation in frequency domain as well as the inverse B-spline neural network model obtained in time domain. Extensive simulation results are included to demonstrate the effectiveness of our nonlinear SC-FDE scheme for Hammerstein channels.

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A practical single-carrier (SC) block transmission with frequency domain equalisation (FDE) system can generally be modelled by the Hammerstein system that includes the nonlinear distortion effects of the high power amplifier (HPA) at transmitter. For such Hammerstein channels, the standard SC-FDE scheme no longer works. We propose a novel Bspline neural network based nonlinear SC-FDE scheme for Hammerstein channels. In particular, we model the nonlinear HPA, which represents the complex-valued static nonlinearity of the Hammerstein channel, by two real-valued B-spline neural networks, one for modelling the nonlinear amplitude response of the HPA and the other for the nonlinear phase response of the HPA. We then develop an efficient alternating least squares algorithm for estimating the parameters of the Hammerstein channel, including the channel impulse response coefficients and the parameters of the two B-spline models. Moreover, we also use another real-valued B-spline neural network to model the inversion of the HPA’s nonlinear amplitude response, and the parameters of this inverting B-spline model can be estimated using the standard least squares algorithm based on the pseudo training data obtained as a byproduct of the Hammerstein channel identification. Equalisation of the SC Hammerstein channel can then be accomplished by the usual one-tap linear equalisation in frequency domain as well as the inverse Bspline neural network model obtained in time domain. The effectiveness of our nonlinear SC-FDE scheme for Hammerstein channels is demonstrated in a simulation study.

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Fluorescence quenching of meso-tetrakis-4-sulfonatophenyl (TPPS4) and meso-tetrakis-4-N-methylpyridil (TMPyP) porphyrins is studied in aqueous solution and upon addition of micelles of sodium dodecylsulfate (SDS), cetyltrimethylammonium chloride (CTAC), N-hexadecyl-N,N-dimethyl-3-ammonio-1-propanesulfonate (HPS) and t-octylphenoxypolyethoxyethanol (Triton X-100). Potassium iodide (KI) was used as quencher. Steady-state Stern-Volmer plots were best fitted by a quadratic equation, including dynamic (K-D) and static (K-s) quenching. Ks was significantly smaller than K-D. Frequency-domain fluorescence lifetimes allowed estimating bimolecular quenching constants, k(q). At 25 degrees C, in aqueous solution, TMPyP shows k(q), values a factor of 2-3 higher than the diffusional limit. TPPS4 shows collisional quenching with pH dependent k(q) values. For TMPyP quenching results are consistent with reported binding constants: a significant reduction of quenching takes place for SDS, a moderate reduction is observed for H PS and almost no change is seen for Triton X-100. Similar data were obtained at 50 C. For CTAC-TPPS4 system an enhancement of quenching was observed as compared to pure buffer. This is probably associated to accumulation of iodide at the cationic micellar interface. The attraction between CTAC headgroups and 1(-), and repulsion between SDS and 1(-), enhances and reduces the fluorescence quenching, respectively, of porphyrins located at the micellar interface. The small quenching of TPPS4 in Triton X-100 is consistent with strong binding as reported in the literature. (C) 2008 Elsevier B.V. All rights reserved.

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Fabric pilling is a serious problem for the apparel industry. Resistance to pilling is normally tested by simulated accelerated wear and manual assessment of degree of pilling based on a visual comparison of the sample to a set of test images. A number of automated systems based on image analysis have been developed. The authors propose new methods of image analysis based on the two-dimensional wavelet transform to objectively measure the pilling intensity in sample images. Initial work employed the detail coefficients of the two-dimensional discrete wavelet transform (2DDWT) as a measure of the pilling intensity of woven/knitted fabrics.

This method is shown to be robust to image translation and brightness variation. Using the approximation coefficients of the 2DDWT, the method is extended to non-woven pilling image sets. Wavelet texture analysis (WTA) combined with principal components analysis are shown to produce a richer texture description of pilling for analysis and classification. Finally, employing the two-dimensional dual-tree complex wavelet transform as the basis for the WTA feature vector is shown to produce good automated classification on a range of standard pilling image sets.

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This paper presents a new time-frequency approach to the underdetermined blind source separation using the parallel factor decomposition of third-order tensors. Without any constraint on the number of active sources at an auto-term time-frequency point, this approach can directly separate the sources as long as the uniqueness condition of parallel factor decomposition is satisfied. Compared with the existing two-stage methods where the mixing matrix should be estimated at first and then used to recover the sources, our approach yields better source separation performance in the presence of noise. Moreover, the mixing matrix can be estimated at the same time of the source separation process. Numerical simulations are presented to show the superior performance of the proposed approach to some of the existing two-stage blind source separation methods that use the time-frequency representation as well.

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A multiple-iteration constrained conjugate gradient (MICCG) algorithm and a single-iteration constrained conjugate gradient (SICCG) algorithm are proposed to realize the widely used frequency-domain minimum-variance-distortionless-response (MVDR) beamformers and the resulting algorithms are applied to speech enhancement. The algorithms are derived based on the Lagrange method and the conjugate gradient techniques. The implementations of the algorithms avoid any form of explicit or implicit autocorrelation matrix inversion. Theoretical analysis establishes formal convergence of the algorithms. Specifically, the MICCG algorithm is developed based on a block adaptation approach and it generates a finite sequence of estimates that converge to the MVDR solution. For limited data records, the estimates of the MICCG algorithm are better than the conventional estimators and equivalent to the auxiliary vector algorithms. The SICCG algorithm is developed based on a continuous adaptation approach with a sample-by-sample updating procedure and the estimates asymptotically converge to the MVDR solution. An illustrative example using synthetic data from a uniform linear array is studied and an evaluation on real data recorded by an acoustic vector sensor array is demonstrated. Performance of the MICCG algorithm and the SICCG algorithm are compared with the state-of-the-art approaches.