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检索条件"任意字段=Advanced Signal Processing Algorithms, Architectures, and Implementations XVII"

共 419 条记录，以下是261-270 订阅

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Optimizing time-frequency distributions for automatic classification

Optimizing time-frequency distributions for automatic classi...

引用

Conference on advanced signal processing - algorithms, architectures, and implementations VII

作者： Atlas, L Droppo, J McLaughlin, J Univ. of Washington (United States) MIT Lincoln Lab. (United States)

ISBN: (纸本)0819425842

An entirely new set of criteria for the design of kernels (generating functions) for time-frequency representations (TFRs) is presented. These criteria aim only to produce kernels (and thus, TFRs) which will enable more accurate classification. We refer to these kernels, which are optimized to discriminate among several classes of signals, as signal class dependent kernels, or simply class dependent kernels. The genesis of the class dependent kernel is to be found in the area of operator theory, which we use to establish a direct link between a discrete-time, discrete-frequency TFR and its corresponding discrete signal. We see that many similarities, but also some important differences, exist between the results of the continuous-time operator approach and our discrete one. The differences between the continuous representations and discrete ones may not be the simple sampling relationship which has often been assumed. From this work, we obtain a very concise, matrix-based expression for a discrete-time/discrete-frequency TFR which is simply the product of the kernel with another matrix. This simple expression opens up the possibility to optimize the kernel in a number of ways. We focus, of course, on optimizations most suitable for classification, and ultimately wind up with the class dependent kernel. When applied to simulated sonar transient signals, we find that our approach does a good job of discriminating within very similar classes of transients and is especially sensitive to differences in time variation across classes.

关键词： Time-frequency analysis Matrices Fourier transforms Convolution Detection theory Distance measurement signal generators signal processing Silicon Stochastic processes

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Low Power implementation of a Turbo-decoder on programmable architectures

Low Power implementation of a Turbo-decoder on programmable ...

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Asia and South Pacific Design Automation Conference (ASP-DAC 2001)

作者： Gilbert, F Worm, A Wehn, N Univ Kaiserslautern Dept Elect Engn & Informat Technol Inst Microelect Syst D-67663 Kaiserslautern Germany

ISBN: (纸本)0780366336

Low Power is an extremely important issue for future mobile radio systems. Channel decoders are essential building blocks of base-band signal processing units in mobile terminal architectures. Thus low power implementations of advanced channel decoding techniques are mandatory. In this paper we present a low power implementation of the most sophisticated channel decoding algorithm (Turbo-decoding) on programmable architectures. Low power optimization is performed on two abstraction levels: on system level by the use of an intelligent cancellation technique, on implementation level by the use of dynamic voltage scaling. With these techniques we can reduce the worst case energy consumption to 55% using data of state-of-the-art processors. Our approach is also applicable for hardware implementations. To the best of our knowledge, this is the first in-depth study of low power implementations of Turbo-decoders based on voltage scheduling for third generation wireless systems.

关键词： Bit error rate Channel coding Dynamic voltage scaling Error correction Forward error correction Iterative decoding Processor scheduling Redundancy signal processing algorithms Turbo codes

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Bandlimits and coherent modulation filtering of non-stationary signals

Bandlimits and coherent modulation filtering of non-stationa...

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Conference on advanced signal processing algorithms, architectures, and Implementantions xviiI

作者： Clark, Pascal Atlas, Les Univ Washington Dept Elect Engn Seattle WA 98195 USA

ISBN: (纸本)9780819472946

Modulation filtering is a technique for filtering slowly-varying envelopes of frequency subbands of a non-stationary signal, ideally without affecting the signal's phase and fine-structure. Coherent modulation filtering is a potentially more effective subtype of such techniques where subband envelopes are determined through demodulation of the subband signal with a coherently detected subband carrier. In this paper we propose a coherent modulation filtering technique based on detecting the instantaneous frequency of a subband from its time-frequency representation. We show that coherent modulation filtering imposes a new bandlimiting constraint on the modulation product plus the ability to recover arbitrarily chosen envelopes and carriers from their modulation product. We show that a carrier estimate based on the time-varying spectral center-of-gravity satisfies the bandlimiting condition as well as Loughlin's previously derived bandlimiting constraint on the instantaneous frequency of carrier. These bandwidth constraints lead to effective and distortion-free modulation filters, offering new approaches for potential signal modification. The spectral center-of-gravity does not satisfy the condition on arbitrary recovery, however, which somewhat limits the flexibility of coherent modulation filtering. Demonstrations are provided with speech signals.

关键词： modulation time-frequency analysis time-varying filters bandwidth speech processing

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signal selection method of a directional diversity for VSB receiver

Signal selection method of a directional diversity for VSB r...

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advanced signal processing algorithms, architectures, and implementations XV

作者： Bae, Jae Hwui Kim, Juyeun Lim, Jong Soo Lee, Soo In Han, Dong Seok Electronics and Telecommunication Research Institute Kyungpook National University

We propose an output signal selection method for the directional diversity in VSB receiver which is to improve the reception performance of VSB system in severe Rayleigh fading channel. The VSB system has only about 0.3% of known training signal for the receiver in a data field and the reception performance of VSB receiver is degraded significantly when there are near-0 dB ghosts in received signal. To overcome this problem, the directional diversity is suggested. In directional diversity the selection of an output signal with best channel condition in point of VSB equalizer is very important to improve VSB reception performance. For the selection of the optimal signal, we extracted channel profiles in time domain for all the signals by correlating the PN511 sequence in VSB field sync and selected one signal by comparing the channel profiles. The simulation results show that the proposed method selects a signal with the best channel condition among the signals, so the reception performance of the VSB system can be improved in severe Rayleigh channels.

关键词： signal receivers

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Analysis of signal reconstruction after modulation filtering

Analysis of signal reconstruction after modulation filtering

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advanced signal processing algorithms, architectures, and implementations XV

作者： Schimmel, Steven M. Atlas, Les E. University of Washington Seattle WA 98195-2500 United States

When the short-time Fourier transform (STFT) of an audio signal is arbitrarily modified, it no longer truly represents a time-domain signal. Classically, the accepted solution to obtain a time-domain signal from a modified STFT (MSTFT) is to invert the MSTFT to a time-domain signal that has an STFT that is closest to the MSTFT in a least squares sense. This is also the approach currently taken by our modulation filtering techniques. However, it was never established that using the original and unmodified STFT phase in this reconstruction is optimal for modulation filtering. In this paper, we compare our signal reconstruction approach to a well-known iterative procedure that approximates a time-domain signal using only the STFT magnitude. We analyze the signal reconstruction of speech signals after filtering them with low-pass, band-pass and high-pass modulation filters. Our study shows that the iterative procedure yields quantitatively and qualitatively comparable signals at significantly higher computational cost. It therefore does not seem a worthwhile alternative to our current reconstruction technique, but it may prove useful for IIR modulation filtering.

关键词： signal reconstruction

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On second-order statistics-based blind equalization of FIR/IIR multiple-input multiple-output channels with common zeros

On second-order statistics-based blind equalization of FIR/I...

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SPIE Conference on advanced signal processing algorithms, architectures, and implementations VIII

作者： Tugnait, JK Huang, B Auburn Univ Dept Elect Engn Auburn AL 36849 USA

ISBN: (纸本)0819429163

The problem of blind equalization of MIMO (multiple-input nultiple-output) communications channels is considered using the second-order statistics of the data. Such models arise when a single receiver data from multiple sources is fractionally sampled (assuming that there is excess bandwidth), or when an antenna array is used with or without fractional sampling. We focus on direct design of finite-length MMSE (minimum mean-square error) blind equalizers. We allow infinite impulse response (IIR) channels. Our approaches also work when the "subchannel" transfer functions have common zeros so long as the common zeros are minimum-phase zeros. We only require that the there exist a causal, stable left inverse (not necessarily unique) to the MIMO transfer function and that the leading coefficient matrix of the MIMO channel impulse response have its rank equal to the number of sources. The channel length or model orders need not be known. The sources are recovered up to a unitary mixing matrix and are further 'unmixed' using higher-order statistics of the data. An illustrative simulation example is provided.

关键词： spatio-temporal processing multiple-input multiple-output channels multiaccess systems blind equalization space diversity time diversity fractional sampling

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Application of symmetric redundant residues for fast and reliable arithmetic

Application of symmetric redundant residues for fast and rel...

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Conference on advanced signal processing algorithms, architectures and implementations XII

作者： Parhami, B Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

ISBN: (纸本)0819445584

Despite difficulties in general division, magnitude comparison, and sign detection, residue number system arithmetic has been used for many special-purpose systems in light of its parallelism and modularity for the most common arithmetic operations of addition/subtraction and multiplication. Computation in RNS requires modular reduction, both for the initial conversion from binary to RNS and after each operation to bring the result back to within a valid residue range. Use of redundant residues simplifies this critical operation, leading to even faster arithmetic. One type of redundant mod-m residue, that keeps the representational redundancy to the minimum of 1 bit per residue, has the nearly symmetric range [-m, m) and allows two values for each pseudoresidue: (m) or (m) - m. We study the extent of simplification and speed-up in the modular reduction process afforded by such redundant residues and discuss its potential implications to the design of RNS arithmetic circuits. In particular, we show that besides cost and performance benefits, introduction of error checking and fault tolerance in arithmetic computations is facilitated when such redundant residues are used.

关键词： computer arithmetic error code modular reduction parallel processing redundant representation RNS

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BLOCK CIRCULANT PRECONDITIONERS FOR 2-D DECONVOLUTION

BLOCK CIRCULANT PRECONDITIONERS FOR 2-D DECONVOLUTION

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CONF ON advanced signal processing algorithms, architectures, AND implementations 3

作者： CHAN, RH NAGY, JG PLEMMONS, RJ Univ. of Hong Kong (Hong Kong China) Univ. of Minnesota (United States) Wake Forest Univ. (United States)

ISBN: (纸本)081940943X

Discretized 2-D deconvolution problems arising, e.g., in image restoration and seismic tomography, can be formulated as 1eas squares compuaions, mm lib— Tx112 where T is often a large-scale rectangular Toeplitz-block matrix. We consider solving such block least squares problems by the preconditioned conjugate gradient algorithm using square nonsingular circulant-block and related preconditioners, constructed from the blocks of the rectangular matrix T. Preconditioning with such matrices allows efficient implementation using the 1-D or 2-D Fast Fourier Transform (FFT). It is well known that the resolution of ill-posed deconvolution problems can be substantially improved by regularization to compensate for their ill-posed nature. We show that regularization can easily be incorporated into our preconditioners, and we report on numerical experiments on a Cray Y-MP. The experiments illustrate good convergence properties of these FET—based preconditioned iterations.

关键词： Image restoration Matrices Deconvolution Fourier transforms Point spread functions Radon Image processing Information operations Iterative methods Mathematics

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Separating desired image and signal invariant components from extraneous variations 6

Separating desired image and signal invariant components fro...

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

作者： Williams, WJ Zalubas, EJ Hero, A UNIV MICHIGAN DEPT ELECT ENGN & COMP SCIANN ARBORMI 48109

ISBN: (纸本)0819422347

Images and signals can be characterized by representations invariant to time shifts, spatial shifts, frequency shifts, and scale changes as the situation dictates. Advances in time-frequency analysis and scale transform techniques have made this possible. The next step is to distinguish between invariant forms representing different classes of image or signal. Unfortunately, additional factors such as noise contamination and `style' differences complicate this. A ready example is found in text, where letters and words may vary in size and position within the image segment being examined. Examples of complicating variations include font used, corruption during fax transmission, and printer characteristics. The solution advanced in this paper is to cast the desired invariants into separate subspaces for each extraneous factor or group of factors. The first goal is to have minimal overlap between these subspaces and the second goal is to be able to identify each subspace accurately. Concepts borrowed from high-resolution spectral analysis, but adapted uniquely to this problem have been found to be useful in this context. Once the pertinent subspace is identified, the recognition of a particular invariant form within this subspace is relatively simple using well-known singular value decomposition techniques.

关键词： scale transform Mellin transform wordspotting singular value decomposition character recognition time-frequency

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Parallel and fault -tolerant techniques for adaptive matched -field processing (MFP) on distributed array systems

Parallel and fault -tolerant techniques for adaptive matched...

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作者： Cho, Kil Seok University of Florida

学位级别：Ph.D.

Sonar array systems in the highly cluttered undersea environment require new and enhanced signal-processing algorithms and sophisticated architectures to meet the stringent real-time and reliability mission requirements. These demands present significant challenges in terms of computational complexity and dependability; and necessitate the development and use of high-performance computing techniques and advanced signal-processing algorithms. Our study introduces novel parallel and fault-tolerant techniques for adaptive matched-field processing (MFP) which we investigated on distributed array systems along with high-performance networking and distributed computing architectures. We developed the parallel algorithms for adaptive MFP to satisfy the computational and memory requirements not readily supported by cutting-edge, high-performance, single-processor systems. We proposed robust MFP algorithms to compensate for the performance degradation presented by sensor failures in sonar array systems. We also developed fault-tolerant parallel algorithms characterized by low overhead and fast recovery time to enhance dependability. We implemented the proposed parallel algorithms and fault-tolerant mechanisms on scalable clusters of computers. We compared their performance considering several critical factors such as parallel and fault-tolerant techniques, communication schemes, and processor architectures. Results show that these parallel and fault-tolerant algorithms can provide a feasible solution for real-time, scalable, and highly reliable beamforming implementations on distributed array systems.

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