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检索条件"任意字段=Conference on Advanced Signal Processing - Algorithms, Architectures, and Implementations VII"

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Modern microphone array for hearing aid and speech processing 6

Modern microphone array for hearing aid and speech processin...

引用

conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Wang, A Yao, K Hudson, RE Korompis, D Lorenzelli, F Soli, SD Gao, S UNIV CALIF LOS ANGELES DEPT ELECT ENGNLOS ANGELESCA 90095

ISBN: (纸本)0819422347

For various audio, teleconference, hearing aid, and voice recognition applications, a microphone array is known to be an effective method to enhance the SNR in noisy environments resulting in significant improvement of speech intelligibility or recognition. We propose an novel electronically steerable microphone array based on the Maximum Energy concentration criterion to form a focused beam toward the desired speech source, attenuating background noises and rejecting discrete spatial interferers. The design and implementation of a prototype DSP-based microphone array system are described. Some details on microphone measurement, calibration, and optimization needed to achieve a high performance microphone array are discussed. Computer simulated and measured array performances are presented to show the effectiveness of the array system.

关键词： Sensors Calibration signal to noise ratio Digital signal processing Computer simulations Near field Acoustics Computing systems signal processing Speaker recognition

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A multivariate speech activity detector based on the syllable rate

A multivariate speech activity detector based on the syllabl...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations viiI

作者： Smith, D Townsend, J Nelson, DJ US Dept Def Ft George G Meade MD 20755 USA

ISBN: (纸本)0819429163

Computationally efficient algorithms which perform speech activity detection have significant potential economic and labor saving benefit, by automating an extremely tedious manual process. In many applications, it is desirable to extract intervals of speech which are bounded by segments of other signal types (fax/modem, music, static, dial tones, etc.). In the past, algorithms which successfully discriminate between speech and one specific other signal type have been developed. Frequently, these algorithms fail when the specific non-speech signal is replaced by a different non-speech signal. Little work has been done on combining such discriminators in order to solve the general speech vs. non-speech discrimination problem. Typically, several signal specific discriminators are blindly combined with predictable negative results. Moreover, when a large number of discriminators are involved, dimension reduction is achieved using Principal Components, which optimally compresses signal variance into the fewest number of dimensions. Unfortunately, these new coordinates are not necessarily optimal for discrimination. In this paper we apply graphical tools to determine a set of discriminators which produce excellent speech vs. non-speech clustering, thereby eliminating the guesswork in selecting good feature vectors. This cluster structure provides a basis for a general multivariate speech vs. non-speech discriminator, which compares very favorably with the TALKATIVE speech extraction algorithm.

关键词： signal processing

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Binary openings and closings in one pass using finite-state machines 6

Binary openings and closings in one pass using finite-state ...

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conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Waltz, FM Consultant (United States)

ISBN: (纸本)0819422347

In a series of ten papers published since 1994, a radically new technique for implementing a wide range of standard image processing operations has been presented, under the acronym SKIPSM (Separated-Kernel Image processing using finite State Machines). Key steps are: (1) the operation is separated into a row operation followed by a column operation, (2) these row and column operations are put in recursive form. That is, in a form compatible with either one-step software implementation or pipelined raster-scan hardware implementation, (3) the resulting operations are realized as FSMs (finite-state machines), and (4) these FSMs are implemented in software or in inexpensive off-the-shelf integrated circuits. Note that this technique does not require separability, in the usual sense. In this paper, the SKIPSM technique is applied to computing binary openings and closings in one pass using arbitrary binary structuring elements. Whether the resulting finite state machines are implemented in software or hardware, the result is generally much faster and/or much cheaper than conventional implementations. Furthermore, this same SKIPSM architecture is highly versatile and programmable, allowing it to be software-reconfigured to perform hundreds of other software- based or pipelined image processing operations, such a binary and grey-level morphology, the Grassfire Transform, binary and grey-level template matching, binary skeletonization, etc.

关键词： fast implementations image processing finite-state machines separability pipelining real-time inspection

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How bad are symmetric Pick matrices

How bad are symmetric Pick matrices

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conference on advanced signal processing algorithms, architectures, and implementations X

作者： Fasino, D Olshevsky, V Univ Udine Dipartimento Matemat & Informat I-33100 Udine Italy

ISBN: (纸本)0819437611

Let P be a symmetric positive definite Pick matrix of order n. The following facts will be proven here: 1. P is the Gram matrix of a set of rational functions, with respect to a inner product defined in terms of a "generating function" associated to P;2. Its condition number is lower-bounded by a function growing exponentially in n. 3. P can be effectively preconditioned by the Pick matrix generated by the same nodes and a constant function.

关键词： Matrices Condition numbers algorithms Computing systems Control systems Image information entropy Linear algebra Mathematics Platinum signal processing

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Performance of space-frequency codes for MIMO OFDM systems

Performance of space-frequency codes for MIMO OFDM systems

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conference on advanced signal processing algorithms architectures and implementations XIII

作者： Wang, JB Yao, K Univ Calif Los Angeles Dept Elect Engn Los Angeles CA 90095 USA

ISBN: (纸本)0819450782

Concatenation of space-time (ST) coding with orthogonal frequency-division multiplexing (OFDM) has gained much interest recently. In this work, we derive the exact pairwise error probability (PEP) of space-frequency (SF) codes for MIMO OFDM Systems. Based on the exact PEP, we derive the tighter upper and lower bounds for the PEP. For asymptotically high SNRs, the design criteria for SF codes differ significantly from those for ST codes over flat fading channels. In this paper, by drawing an analogy between SF and ST codes, we show that when the number of receive antennas is large, the minimum Euclidean distance among code words dominates the performance of SF codes. Therefore, SF codes can be optimized by using the Euclidean-distance criterion valid for AWGN channels. Simulation results are given to show that the results valid for a number of receive antennas tending to infinity still provide correct indications when the number of antennas is small.

关键词： Orthogonal frequency division multiplexing Antennas Modulation Lutetium signal to noise ratio Computer simulations Demodulation signal processing

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Performance of coherent adaptation algorithms for the WCDMA smart antenna system in Rayleigh flat-fading channel

Performance of coherent adaptation algorithms for the WCDMA ...

引用

conference on advanced signal processing algorithms, architectures, and implementations XI

作者： Kong, DG Lee, K Chun, J Choi, P Yun, JC Lee, SJ Lee, JH Samsung Adv Inst Technol Human Comp Interact Lab Suwon South Korea

ISBN: (纸本)0819441880

Coherent adaptation algorithms are proposed for a beamformer in the WCDMA reverse link. The coherent adaptation algorithm improves the convergence speed by constraining the desired signal component of the filtered output to be always coherent in phase with the reference signal at each iteration. With the coherent constraint, the performance of coherent methods is significantly improved. We present the simulation results that show the various desirable characteristics of coherent methods such as a fast convergence speed. insensitiveness to fading parameters over conventional methods.

关键词： coherent adaptation algorithm smart antenna WCDMA array signal processing

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Spatial polarimetric time-frequency distributions and applications to direction-of-arrival estimation

Spatial polarimetric time-frequency distributions and applic...

引用

conference on advanced signal processing algorithms architectures and implementations XIII

作者： Zhang, YM Amin, MG Obeidat, BA Villanova Univ Ctr Adv Commun Villanova PA 19085 USA

ISBN: (纸本)0819450782

Time-frequency distributions (TFDs) have evolved to be a powerful technique for nonstationary signal analysis and synthesis. With the use of a multi-sensor array, spatial time-frequency distributions (STFDs) have been developed and successfully applied to high-resolution direction-of-arrival (DOA) estimation and blind recovery of the source waveforms. In this paper, the polarimetric dimension is introduced to the STFDs resulting in the spatial polarimetric time-frequency distributions (SPTFDs) as a platform for the processing of non-stationary polarized signals. In the SPTFD platform, polarized signals are decomposed (projected) into two orthogonal polarization components, such as horizontal and vertical, and later processed in a manner where their polarization characteristics are exploited. This empowers the STFDs with additional degrees of freedom and improves the robustness of the signal and noise subspaces, and therefore, serving to enhance DOA estimation, signal recovery, and source separation performance. To demonstrate the advantages of the SPTFDs, the polarimetric time-frequency MUSIC (PTF-MUSIC) method for DOA estimation is proposed based on the SPTFD platform and is shown to outperform the time-frequency, polarimetric, and conventional MUSIC methods.

关键词： time-frequency distributions array signal processing polarization smart antennas MUSIC

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Image processing and low discrepancy sequences

Image processing and low discrepancy sequences

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conference on advanced signal processing algorithms, architectures, and implementations IX

作者： Nair, D Wenzel, L Natl Instruments Austin TX 78727 USA

ISBN: (纸本)0819432938

It is well known that high-dimensional integrals can be solved with Monte Carlo algorithms. Recently, it was discovered that there is a relationship between low discrepancy sets and the efficient evaluation of higher-dimensional integrals. Theory suggests that for midsize dimensional problems, algorithms based on low discrepancy sets should out perform all other existing methods by an order of magnitude in terms of the number of sample points used to evaluate the integrals. We show that the field of image processing can potentially take advantage of specific properties of low discrepancy sets. To illustrate this, we applied the theory of low discrepancy sequences to some relatively simple image processing and computer vision related operations such as the estimation of gray level image statistics, fast location of objects in a binary image and the reconstruction of images from a sparse set of points. Our experiments show that compared to standard methods, the proposed new algorithms are faster and statistically more robust. Classical low discrepancy sets based on the Halton and Sobol' sequences were investigated thoroughly and showed promising results. The use of low discrepancy sequences in image processing for image characterization, understanding and object recognition is a novel and promising area for further investigation.

关键词： sampling random sampling low-discrepancy sequences image statistics image reconstruction

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Approximation by structured lower rank matrices

Approximation by structured lower rank matrices

引用

SPIE conference on advanced signal processing algorithms, architectures, and implementations viiI

作者： Chu, MT Funderlic, RE Plemmons, RJ N Carolina State Univ Dept Math Raleigh NC 27695 USA

ISBN: (纸本)0819429163

We consider two general procedures for constructing the nearest approximation of a given matrix by one with any lower rank and any linear structure. Nearness can be measured in any matrix norm. Structured low rank matrices arise in various applications, including image enhancement and model reduction. In practice, the empirical data collected in the matrix often either do not maintain the specified structure or do not induce the desirable rank. It is an important task to search for the nearest structured lower rank approximation of a given matrix. The techniques developed in this paper can easily be implemented for numerical computation. In particular, it is shown that the computations can be approached using efficient optimization packages. The special case of Toeplitz structure using the Frobenius matrix norm is discussed in detail to illustrate the ideas, and numerical tests are reported. The procedures developed herein can be generalized to consider a much broader range of approximation problems.

关键词： Matrices signal processing MATLAB Optimization (mathematics) Image enhancement Computer programming Computer science Fourier transforms Image processing Mathematics

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A spectral phase algorithm for detecting and estimating pitch

A spectral phase algorithm for detecting and estimating pitc...

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conference on advanced signal processing algorithms, architectures, and implementations XI

作者： Tempkin, JA Nelson, DJ USA Dept Def Ft George G Meade MD 20755 USA

ISBN: (纸本)0819441880

We describe a fast and efficient algorithm for automatic detection and estimation of the fundamental frequency F0 of a harmonic time-domain signal. The method is based on differentiation of the short time Fourier transform (STFT) phase, which is implemented as a cross-spectral product. In estimating and isolating the fundamental frequency, several enhancement processes are developed and applied to the TF surface to improve the signal quality. We describe the algorithm in detail and demonstrate the processing gain achieved at each step. In addition, we apply the algorithm to human speech to recover the pitch fundamental F0 and report the evaluation of the algorithm's performance on the Western Michigan vowel corpus [3].

关键词： signal processing

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