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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...

引用

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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A DSP implementation of source location using microphone arrays 6

A DSP implementation of source location using microphone arr...

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

作者： Rabinkin, DV Renomeron, RJ Dahl, A French, JC Flanagan, JL Bianchi, MH RUTGERS STATE UNIV CTR COMP AIDS IND PRODPISCATAWAYNJ 08855

ISBN: (纸本)0819422347

The design, implementation, and performance of a low-cost, real-time DSP system for source location is discussed. The system consists of an 8-element electret microphone array connected to a signalogic DSP daughterboard hosted by a PC. The system determines the location of a speaker in the audience in an irregularly shaped auditorium. The auditorium presents a non-ideal acoustical environment;some of the walls are acoustically treated, but there still exists significant reverberation and a large amount of low frequency noise from fans in the ceiling. The source location algorithm is implemented in a two step process: The first step determines time delay of arrival (TDOA) for select microphone pairs. A modified version of the Cross-Power Spectrum Phase Method is used to compute TDOAs and is implemented on the DSP daughterboard. The second step uses the computed TDOAs in a least mean squares gradient descent search algorithm implemented on the PC to compute a location estimate. © 1996 SPIE. All rights reserved.

关键词： microphone arrays speech source location real-time processing

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Some significant communications processing realized with adaptive Bragg gratings

Some significant communications processing realized with ada...

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

作者： Upton, E Wickham, M Smith, M Hart, J Flynn, R Havstad, S Nachman, P TRW Co Inc Redondo Beach CA 90278 USA

ISBN: (纸本)0819429163

Higher data capacity demands and lower interference requirements in the wireless communications arena are exploiting higher carrier frequencies and wider modulation bandwidths. Circuitry which can perform intermediate frequency processing over these more demanding ranges is needed to provide complex signal processing without commercial penalties. Photonics technologies utilizing Bragg Grating signal processing (BGSP) can bridge the gap between the very high frequency RF millimeter wave integrated circuit (MIMIC) domains at the antenna interface and the CMOS digital signal processors at the base band frequency interface. The desirable benefits of multiple tap adaptive finite impulse response (FIR) and infinite impulse response (IIR) filters and equalizers are well known;however, they are usually the province of digital signal processing and force the sample rates prior to these processors to a higher overall system power consumption level. BGSP provides these functions with discrete taps and digital controls but at the bandwidths usually reserved for RF circuitry because the actual processing occurs at optical frequencies and at wave lengths which are compatible with integrated circuit technologies. The high performance benefits of photonic processing can be realized if the stability control of the Bragg grating is derived from the same metric which induces in photonics its sensitivity to drift. We will present a orthogonally coded tap modulation technique which stabilizes the transfer function of the signal processor and enables significant adaptive IF signal processing to be obtained with very low size, weight, and power. Our demonstration of a photonic proof-of-concept architecture is a reconfigurable multiple tap FIR filter that is dynamically controlled to perform low pass, high pass, band pass, and band stop filters operating over bandwidths of 3 GHz.

关键词： signal processing Fiber Bragg gratings Filtering (signal processing) Control systems Linear filtering Digital signal processing Optical filters Servomechanisms Modulators Electronic filtering

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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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Approach to the joint optimization of video communications and video signal processing

Approach to the joint optimization of video communications a...

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

作者： Pearlman, WA Medley, MJ Suter, BW Rensselaer Polytech Inst Ctr Next Generat Video Res Troy NY 12180 USA

ISBN: (纸本)0819441880

Compressed video bitstreams require protection from channel errors in a wireless channel. The three-dimensional (3-D) SPIHT coder has proved its efficiency and its real-time capability in compression of video. A forward-error-correcting (FEC) channel (RCPC) code combined with a single ARQ (automatic-repeat-request) proved to be an effective means for protecting the bitstream. In this paper, the need for ARQ is eliminated by making the 3-D SPIHT bitstream more robust and resistant to channel errors. Packetization of the bitstream and the reorganization of these packets to achieve scalability in bit rate and/or resolution in addition to robustness is demonstrated and combined with channel coding to not only protect the integrity of the packets, but also allow detection of packet decoding failures, so that only the cleanly recovered packets are reconstructed. In extensive comparative tests, the reconstructed video is shown to be superior to that of MPEG-2, with the margin of superiority growing substantially as the channel becomes noisier.

关键词： video compression video transmission robust source coding error resilient transmission combined source-channel coding 3-D wavelet transform embedded wavelet coding

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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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Optimizing time-frequency distributions for automatic classification

Optimizing time-frequency distributions for automatic classi...

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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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New one-hot RNS structures for high-speed signal processing

New one-hot RNS structures for high-speed signal processing

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conference on advanced signal processing algorithms, architectures and implementations XII

作者： Conway, R Conway, T Nelson, J Univ Limerick ECE Dept Limerick Ireland

ISBN: (纸本)0819445584

New efficient structures using the one-hot residue number system (OHRNS) are presented. Normally the RNS uses a binary representation for the residues, though recently there has been renewed interest in the OHRNS, which uses a simple, but novel representation for the residues. The basic component of the OHRNS is the barrel shifter, making the OHRNS suitable for very high speed applications. The first of the new structures presented reduces the power dissipation in OHRNS adder trees. A modification to the normal barrel shifter is proposed, which reduces the power dissipated by as much as 30%. This improvement is obtained through the use of the modified barrel shifter and the appropriate connection of active-low and active-high stages. This overall power reduction offers the possibility of using the OHRNS in place of a typical full adder based tree in high speed DSP applications. A new storage register for one-hot representations is detailed, which overcomes the problem of having to use a large number of registers. A new architecture is presented for fast OHRNS sign detection. Sign detection is complex and slow to perform in the RNS. A mixed radix conversion (MRC) is typically used for sign detection in the OHRNS. The new sign detection architecture is based on a new property of the Chinese Remainder Theorem (CRT) and is significantly faster than the MRC approach for large moduli sets. Simulation results using SPICE are detailed for the new structures.

关键词： RNS one-hot sign detection

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3D Object Matching on the GPU Using Spin-Image Surface Matching

3D Object Matching on the GPU Using Spin-Image Surface Match...

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conference on advanced signal processing algorithms, architectures, and Implementantions XVIII

作者： Davis, Nolan Braunreiter, Dennis Tebcherani, Cezario Tanida, Masatoshi Sci Applicat Int Corp San Diego CA 92121 USA

ISBN: (纸本)9780819472946

Spin-image surface matching is a technique for locating objects in a scene by processing three-dimensional surface information from sources such as light detection and ranging (LIDAR), structured light photography, and tomography. It is attractive for parallel processing on graphics processing units (GPUs) because the two main computational steps matching pairs of spin-images by correlation, and matching pairs of points between model and scene - are explicitly parallel. By implementing these parallel computations on the GPU, as well as recasting serial portions of the algorithm into a parallel form and structuring the algorithm to limit data exchanges between host and GPU, this project achieved an overall speedup of 20 times or more compared to conventional serial processing. A demonstration application has been developed that allows users to select among a set of models and scenes and then applies the spin-image surface matching algorithm to match the selected models to the scene. It also has several user interface controls for changing parameters. One new parameter is a geometric consistency ratio (GCR) that quantifies the matching performance and provides a measure for discarding low-quality matches. By toggling between GPU- and host-based processing, the application demonstrates the speedup achieved with parallelization on the GPU.

关键词： spin image surface matching object matching image correlation geometric consistency GPU graphical processing unit parallel processing speedup

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