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Signal Processing Letters, IEEE

Issue 6 • Date June 2004

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Displaying Results 1 - 19 of 19
  • Table of contents

    Publication Year: 2004 , Page(s): c1 - c4
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    Freely Available from IEEE
  • IEEE Signal Processing Letters publication information

    Publication Year: 2004 , Page(s): c2
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  • Anti-jamming filtering in the autocorrelation domain

    Publication Year: 2004 , Page(s): 525 - 528
    Cited by:  Papers (7)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (144 KB) |  | HTML iconHTML  

    An anti-jamming filtering technique is presented that it fully eliminates the jamming (or noncooperative) signal and its performance is noise-independent. This technique uses multiple receivers; a filtering process explores the nonoverlapping properties of the signals in the autocorrelation domain; and a matching of the input and the output statistics. A limited simulation study supports the theory. View full abstract»

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  • Exact fractional-order differentiators for polynomial signals

    Publication Year: 2004 , Page(s): 529 - 532
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (176 KB) |  | HTML iconHTML  

    A discrete-time fractional-order differentiator is modeled as a finite-impulse response (FIR) system. The system yields fractional-order derivatives of Riemann-Liouville type for a uniformly sampled polynomial signal. The computation of the output signal is based on the additive combination of the weighted outputs of N cascaded first-order digital differentiators. For differentiators of fractional order with a terminal value equal to zero, the weights are time-varying. The weights are obtained in a closed form involving the Stirling numbers of the first kind. The system tends to a time-invariant integer-order differentiator when the order of the derivative tends to an integer value. It yields exact fractional- or integer-order derivatives of a sampled polynomial signal of a certain order. View full abstract»

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  • Flexible parametrization of postnonlinear mixtures model in blind sources separation

    Publication Year: 2004 , Page(s): 533 - 536
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB) |  | HTML iconHTML  

    This letter introduces a flexible parametrization for the nonlinear compensator in the postnonlinear mixture model. The idea is to parametrize not the compensator but a set of quantiles of the distribution of its output. They are specified by the density-quantiles which are parametrized through a spline basis. Good results can be achieved with only a few splines. View full abstract»

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  • Fast algorithms for blind estimation of reverberation time

    Publication Year: 2004 , Page(s): 537 - 540
    Cited by:  Papers (11)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB) |  | HTML iconHTML  

    The reverberation time (RT) measures the persistence of a sound in enclosed acoustic spaces. In a previous work, a method for the blind maximum-likelihood estimation (MLE) of RT using passively received microphone signals was presented. The procedure overcomes the drawbacks of current methods that use a controlled sound source for RT determination. Here, fast algorithms for online implementation of the method are developed. One algorithm, suitable for a one-time determination of the RT, requires O(N) computations for a data frame of length N. A second IIR algorithm, based on Q-levels of quantization, requires O(Q) computations. Results for speech data and choice of algorithms are discussed. View full abstract»

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  • Multicriteria design of oversampled uniform DFT filter banks

    Publication Year: 2004 , Page(s): 541 - 544
    Cited by:  Papers (17)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (216 KB) |  | HTML iconHTML  

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. However, subband processing causes signal degradations due to aliasing effects and amplitude distortions. This problem is unavoidable due to further filtering operations in subbands. In this letter, the problems of aliasing effect and amplitude distortion are studied. Prototype filters which are optimized with respect to those properties are designed and their performances are compared. Moreover, the effect of the number of subbands, the oversampling factors and the length of the prototype filter are also studied. Using the multicriteria formulation, all Pareto optimums are sought via the nonlinear programming technique. We find that the prototype filter designed via the Kaiser window provides the best overall performance among the methods we studied. Also, there is a critical oversampling factor beyond which the improvement of performance is diminishing. Finally, if the length of the prototype filter increases with the number of subbands, an increase in the number of subbands will not deteriorate the performance. View full abstract»

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  • A new class of shift-invariant operators

    Publication Year: 2004 , Page(s): 545 - 548
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (160 KB) |  | HTML iconHTML  

    This letter proposes a class of operators with a shift invariance property. These operators are derived from two-dimensional (2-D) complex moment invariants based on the observation that there is a duality between rotation invariance and shift invariance. A general form of the shift invariants belonging to this class is presented, which shows that polyspectral invariants such as the power spectrum and the bispectrum are members of the class. Methods for computing shift invariants for one-dimensional (1-D) and 2-D signals are also presented. The examples given in the paper suggest that the higher order operators can preserve the original signal waveform better than autocorrelation. View full abstract»

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  • Generalized cross validation for multiwavelet shrinkage

    Publication Year: 2004 , Page(s): 549 - 552
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB) |  | HTML iconHTML  

    Traditional multiwavelet shrinkage denoising techniques require a priori knowledge of noise variance that may not be obtained in some practical situations. By using generalized cross validation (GCV), we propose in this paper a new level-dependent risk estimator for multiwavelet shrinkage that does not require such a priori information. Simulation results verify that the resulted risk estimator gives better indication on threshold selection comparing with the traditional GCV method. Improved denoising performance is then achieved particularly for higher multiplicity multiwavelet shrinkage. View full abstract»

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  • On the use of phase and energy for musical onset detection in the complex domain

    Publication Year: 2004 , Page(s): 553 - 556
    Cited by:  Papers (33)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (192 KB) |  | HTML iconHTML  

    We present a study on the combined use of energy and phase information for the detection of onsets in musical signals. The resulting method improves upon both energy-based and phase-based approaches. The detection function, generated from the analysis of the signal in the complex frequency domain is sharp at the position of onsets and smooth everywhere else. Results on a database of recordings show high detection rates for low rates of errors. The approach is more robust than its predecessors both theoretically and practically. View full abstract»

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  • Adaptive filtering with averaging-based algorithm for feedforward active noise control systems

    Publication Year: 2004 , Page(s): 557 - 560
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    This letter proposes an adaptive filtering with averaging-based algorithm for active noise control (ANC) systems. This algorithm uses a similar structure as that of the FxLMS-based ANC system. The proposed algorithm, called Filtered-x Adaptive Filtering with Averaging (FxAFA) algorithm, uses averages of both data and correction terms to find the updated values of the tap weights of the ANC controller. The computer simulations are conducted for single-channel feedforward ANC systems. It is shown that the proposed algorithm gives fast convergence as compared with the FxLMS algorithm and achieves better performance in the presence of the measurement noise. The comparison with the FxRLS algorithm shows that the proposed FxAFA algorithm is a better choice for low computational complexity and stable performance. View full abstract»

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  • Fade-resistant transmission over time-varying wireless channels

    Publication Year: 2004 , Page(s): 561 - 564
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (144 KB) |  | HTML iconHTML  

    Performance of a parallel symbol transmission scheme is analyzed in time-varying Nakagami fading channels. We show that instantaneous signal-to-interference-plus-noise ratio of the parallel symbol transmission scheme at the matched-filter output is higher than that of a conventional system. Furthermore, the performance gain of the parallel symbol transmission scheme over the conventional system increases as the number of independent fades within the frame increases and/or the Nakagami parameter m decreases. View full abstract»

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  • Log likelihood ratio calculation without SNR estimation for forward link DS-CDMA receivers

    Publication Year: 2004 , Page(s): 565 - 568
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (120 KB) |  | HTML iconHTML  

    This letter deals with the log likelihood ratio computation for the turbo decoder operating on the forward link of a CDMA system. It is shown that the log likelihood ratio contains a time-varying factor, which is the power ratio between the data and the pilot channels. Two alternative techniques for estimating the power ratio are compared along with an ideal scheme. View full abstract»

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  • TCP retransmission timeout algorithm using weighted medians

    Publication Year: 2004 , Page(s): 569 - 572
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (200 KB) |  | HTML iconHTML  

    This letter presents a new retransmission timeout (RTO) algorithm based on recursive weighted median (RWM) filters for the transmission control protocol (TCP). The RTO algorithm utilized in current TCP implementations is Jacobson's algorithm, which is based on recursive linear filtering. While linear filters are adequate for estimation in Gaussian signal environments, the round trip time (RTT) signals filtered to determine the RTOs are often impulsive. Thus, Jacobson's algorithm is not effective in many cases. The proposed algorithm employs RWM filters that yield improved performance when operating on RTT signals with heavy tailed statistics. Simulation results show that the proposed method yields significantly tighter RTT bounds than Jacobson's method over Internet traffic with heavy tailed statistics. View full abstract»

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  • Multiple antenna transmission with channel state information: a low-rate feedback approach

    Publication Year: 2004 , Page(s): 573 - 576
    Cited by:  Papers (2)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (168 KB) |  | HTML iconHTML  

    This paper presents a novel multiple antenna transmission scheme that depends on channel state information. The proposed approach uses low-rate feedback to adaptively estimate the downlink fading channel at the base-station. The optimal transmission scheme given channel state information can in turn be determined from these channel estimates. It is shown that the channel estimation requires selective feedback of the adaptive filter prediction errors, that can be computed at the mobile by receiving a beamformed pilot signal. To this end, a randomized channel inversion pilot beamforming scheme is developed that makes possible this feedback of prediction errors possible. Further, this method is shown to be relatively insensitive to feedback quantization effects for a given average feedback rate. Simulation results indicate that the feedback scheme provides significant closed-loop beamforming gains with low bit rate feedback even for large number of transmit antennas. View full abstract»

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  • An edge-oriented spatial interpolation for consecutive block error concealment

    Publication Year: 2004 , Page(s): 577 - 580
    Cited by:  Papers (33)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (368 KB) |  | HTML iconHTML  

    The coding scheme is currently the most commonly used process of compressing image data. However, this process often results in very serious distortion in the decoded image if the bit-stream suffers from damages. In order to address this drawback, this letter aims to develop a new error concealment technique based on edge-oriented interpolation for still image or intra-frame correction. The first step involves finding the edge direction of a lost block by using one-dimensional matching techniques from two boundaries of neighboring blocks. Then, the error pixels are recovered by weighting linear interpolation along the estimated edge direction. Afterwards, the median filter is used to recover residual damaged-pixels. Simulations demonstrate that the important edge information can be significantly recovered, and also prove that this method performs better compared to other methods that use both subjective and objective measures. View full abstract»

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  • On the use of channel-attentive MFCC for robust recognition of partially corrupted speech

    Publication Year: 2004 , Page(s): 581 - 584
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB)  

    This letter proposes a channel-attentive mel frequency cepstral coefficient (CAMFCC) method to improve the utilization of uncorrupted or more reliable frequency bands for robust speech recognition. This method obtains a channel attention matrix by reliability estimation of mel filter bank channels, and both the input mel frequency cepstral coefficients and the mean vectors of hidden Markov models are corrected using the channel attention matrix at the output probability calculation of the Viterbi decoding. Experimental results on the TIDIGITS database corrupted by various band-selective noises indicated that the proposed CAMFCC method utilizes the uncorrupted partial frequency bands better than a multiband method, resolving the limitation of noise localization caused by the fixed boundaries of the multiband approach. View full abstract»

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  • Suppression of additive noise using a power spectral density MMSE estimator

    Publication Year: 2004 , Page(s): 585 - 588
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (168 KB) |  | HTML iconHTML  

    In this letter, we propose a novel speech enhancement approach, called power spectral density minimum mean-square error (PSD-MMSE) estimation-based speech enhancement, which is implemented in the power spectral domain where stationary stochastic noise can be modeled as the exponential distribution. Speech magnitude-squared spectra are modeled as the mixed exponential distribution. And an MMSE estimator is constructed based on the parametric distributions. Besides, a fast algorithm is presented to implement the approach in real time. Experimental results of Itakura-Saito distortion measures show that the proposed approach is superior to alternative speech enhancement algorithms. View full abstract»

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  • IEEE Signal Processing Society Information

    Publication Year: 2004 , Page(s): c3
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    Freely Available from IEEE

Aims & Scope

The IEEE Signal Processing Letters is a monthly, archival publication designed to provide rapid dissemination of original, cutting-edge ideas and timely, significant contributions in signal, image, speech, language and audio processing.

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Meet Our Editors

Editor-in-Chief
Peter Willett
University of Connecticut
Storrs, CT 06269
peter.willett@uconn.edu