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Digital Signal Processing Workshop and 5th IEEE Signal Processing Education Workshop, 2009. DSP/SPE 2009. IEEE 13th

Date 4-7 Jan. 2009

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Displaying Results 1 - 25 of 152
  • [Front cover]

    Publication Year: 2009, Page(s): 1
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  • [Breaker page]

    Publication Year: 2009, Page(s): 1
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  • [Breaker page]

    Publication Year: 2009, Page(s): ii
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  • Contributor listings

    Publication Year: 2009, Page(s): iii
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  • [Breaker page]

    Publication Year: 2009, Page(s): iv
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  • Table of contents

    Publication Year: 2009, Page(s):v - xviii
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  • Haar-Like Filtering with Center-Clipped Emphasis for Speech Detection in Sensornet

    Publication Year: 2009, Page(s):1 - 4
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (4114 KB) | HTML iconHTML

    The use of Haar-like filtering for resourced-constrained speech detection in sensornet application is explored. The simple Haar-like filters having variable filter width and shift width are trained to learn appropriate filter parameters from the training samples to detect speech. To further refine the accuracy, the center-clipped emphasis is proposed as a new degree of freedom for more adaptive Ha... View full abstract»

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  • A Kalman Filter based Fast Noise Suppression Algorithm

    Publication Year: 2009, Page(s):5 - 9
    Cited by:  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1557 KB) | HTML iconHTML

    We have proposed a robust noise suppression algorithm with Kalman filter theory [7]. In this paper, we propose a fast noise suppression algorithm by modifying the canonical state space model in [7]. The algorithm aims to achieve robust noise suppression with reduced computational complexity without sacrificing high quality of speech signal. The remarkable features of the proposed algorithm are tha... View full abstract»

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  • Multiband Excitation for Speech Enhancement

    Publication Year: 2009, Page(s):10 - 15
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (190 KB) | HTML iconHTML

    The speech enhancement algorithm proposed aims to improve the quality of denoised speech by introducing voicing information to a Wiener-type spectral amplitude gain function. A constrained multiband excitation (MBE) model is used to emphasize harmonic components of the glottal input; and a low variance and bias autoregressive multitaper (ARMT) estimate models the vocal tract. Objective and subject... View full abstract»

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  • Distributed Audio Coding with Efficient Source Correlation Extraction

    Publication Year: 2009, Page(s):16 - 20
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (266 KB) | HTML iconHTML

    Distributed source coding is one of the enabling technologies for efficient bandwidth utilization in wireless sensor networks and is consequently of great current interest. This paper studies its application to audio signals, using a transform weighted interleaved vector quantization (TWIN-VQ) framework and allowing a sensor node to passively receive and use information from neighboring sensors th... View full abstract»

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  • A Temporally Varying Objective Audio Quality Metric

    Publication Year: 2009, Page(s):21 - 25
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (103 KB) | HTML iconHTML

    Currently, objective audio quality measures designed to mimic the human psychoacoustic system operate by giving a single number to designate the quality of an entire sequence. This methodology is only effective when both the error and the underlying audio sequence are stationary. It is highly desirable to have a quality measure that can predict the quality of time varying errors and sequences. Thi... View full abstract»

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  • Spectral Multi-Scale Analysis for Multi-Pitch Tracking

    Publication Year: 2009, Page(s):26 - 31
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (4713 KB) | HTML iconHTML

    This paper proposes a robust and accurate multi-pitch estimation method for multiple voices. This method is based on the spectral analysis of the mixture sound multi-scale product. The multi-scale product (PM) consists of making the product of wavelet transform coefficients. The wavelet used is the quadratic spline function. Simulation results showed that the proposed method can robustly estimate ... View full abstract»

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  • A Low Complexity Noise Suppressor with Hybrid Filterbanks and Adaptive Time-Frequency Tiling

    Publication Year: 2009, Page(s):32 - 36
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (5066 KB) | HTML iconHTML

    This paper proposes a low complexity noise suppressor with hybrid filterbanks and adaptive time-frequency tiling. An analysis hybrid filterbank provides efficient transformation by further decomposing low-frequency bins after a coarse transformation with a short frame size. A synthesis hybrid filterbank also reduces computational complexity in a similar fashion to the analysis hybrid filterbank. A... View full abstract»

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  • Speaker Identification in Room Reverberation Using GMM-UBM

    Publication Year: 2009, Page(s):37 - 41
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (155 KB) | HTML iconHTML

    Speaker recognition systems tend to degrade if the training and testing conditions differ significantly. Such situations may arise due to the use of different microphones, telephone and mobile handsets or different acoustic conditions. Recently, the effect of the room acoustics on speaker identification (SI) has been investigated and it has been shown that a loss in accuracy results when using cle... View full abstract»

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  • Comparison of Localization Algorithms Using Attenuation Estimates

    Publication Year: 2009, Page(s):42 - 47
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (164 KB) | HTML iconHTML

    In this paper, algorithms for locating sources using attenuation estimates are discussed. These algorithms assume that the sensor locations are known. Six closed form methods and an iterative method are described. A spatial histogram algorithm is introduced. The computational complexities of all the algorithms are discussed and the accuracies of the algorithms are compared as a function of error i... View full abstract»

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  • Learning the Intrinsic Dimensions of the Timit Speech Database with Maximum Variance Unfolding

    Publication Year: 2009, Page(s):48 - 53
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (511 KB) | HTML iconHTML

    Modern methods for nonlinear dimensionality reduction have been used extensively in the machine learning community for discovering the intrinsic dimension of several datasets. In this paper we apply one of the most successful ones maximum variance unfolding on a big sample of the well known speech benchmark TIMIT. Although MVU is not generally scalable, we managed to apply to 1 million 39-dimensio... View full abstract»

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  • An Index to Measure "Transient-Ness" of Speech

    Publication Year: 2009, Page(s):54 - 59
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (6591 KB) | HTML iconHTML

    Several studies have shown that emphasis of speech transients can improve the intelligibility of speech in noise. However, each study used a different method to define speech transients, and comparisons across methods are difficult. This paper introduces an index to quantify the transient nature of speech (specifically, the extent to which the onsets and offsets of formants are emphasized compared... View full abstract»

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  • Capon Beamforming for Active Ultrasound Imaging Systems

    Publication Year: 2009, Page(s):60 - 65
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (234 KB) | HTML iconHTML

    Medical ultrasound imaging has unique requirements regarding spatial and amplitude resolution, near-field focusing, wide bandwidth, and real-time operation. Only recently has Capon beamforming been adapted to this. We give examples of images of point targets, cysts, and regions dominated by speckle and discuss how subaperture smoothing, diagonal loading and range/time averaging must be balanced fo... View full abstract»

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  • Closed-Form MSE Performance of the Distributed LMS Algorithm

    Publication Year: 2009, Page(s):66 - 71
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (6164 KB) | HTML iconHTML

    Mean-square error (MSE) performance analysis is conducted for a novel distributed least-mean square (D-LMS) algorithm, which is based on consensus, in-network, adaptive estimation using wireless sensor networks (WSNs). For sensor observations that are linearly related to the time-invariant parameter of interest and independent Gaussian data, exact closed-form expressions are derived for the global... View full abstract»

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  • A Simplified Predistorter for Distortion Compensation of Parallel Wiener-Type Systems Based on Direct Learning Architecture

    Publication Year: 2009, Page(s):72 - 77
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (146 KB) | HTML iconHTML

    Predistortion of parallel wiener-type systems is considered in this paper. The predistorter is first modeled as a Volterra series. In order to reduce computation complexity, a simplified predistorter model constructed using adaptive FIR filters is proposed. The coefficients of these two suggested predistorters are estimated using the nonlinear filtered-x least mean squares (NFtimesLMS) algorithm b... View full abstract»

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  • Stochastic Search Methods to Improve the Convergence of Adaptive Notch Filters

    Publication Year: 2009, Page(s):78 - 83
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (399 KB) | HTML iconHTML

    Adaptive notch filters (ANFs) are known to have convergence problems due to their non-quadratic error surface. We propose two approaches to improve the convergence of the ANF. The first approach is based on the method of stochastic search. The second approach checks to see whether the estimated signal is correlated to the measurement or is just filtered white noise. The ANF is reinitialized when t... View full abstract»

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  • Fundamental Issues in the Stability of Adaptive IIR Filters

    Publication Year: 2009, Page(s):84 - 89
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (125 KB) | HTML iconHTML

    Adaptive IIR filter analysis is more complicated than for the FIR case because (a) some algorithm signals are generated by the adaptive filter itself, and (b) the prediction error relates to the adapted parameters via a filtering operation. Averaging analyses of stability address the first issue by linearization about the convergence point, and the second by using passivity of the error operator. ... View full abstract»

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  • Block Adaptive ICA with a Time Varying Mixing Matrix

    Publication Year: 2009, Page(s):90 - 95
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1027 KB) | HTML iconHTML

    Independent component analysis effectively resolves fixed linear combinations of independent distributions under requirements that include long block length. In environments where the rules of source combination change rapidly, adaptive or block adaptive methods must be deployed; and associated problems of convergence and permutation ambiguity solved. We propose using ICA on overlapping blocks and... View full abstract»

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  • Image Interpolation Exploiting Phase Diversity

    Publication Year: 2009, Page(s):96 - 100
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (322 KB) | HTML iconHTML

    In this paper we present an analysis-synthesis system with adaptive synthesis filters. The general formulation can be employed for image compression applications, but can also be used for image size enlargement. The synthesis section is constructed to contain a multiplicity of outputs with different phase characteristics. By exploiting phase diversity in synthesis, improvement in both PSNR and sub... View full abstract»

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  • Mean and Mean-Square Analysis of the Complex LMS Algorithm for Non-Circular Gaussian Signals

    Publication Year: 2009, Page(s):101 - 106
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (5118 KB) | HTML iconHTML

    The least-mean-square (LMS) algorithm is a useful and popular procedure for adaptive signal processing of both real-valued and complex-valued signals. Past analysis of the complex LMS algorithm has assumed that the input signal vector is circularly-distributed, such that the pseudo-covariance matrix of the input signal is zero. In this paper, we relax this assumption, providing a complete mean and... View full abstract»

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