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检索条件"任意字段=Advanced Signal Processing Algorithms, Architectures, and Implementations 1990"

共 433 条记录，以下是211-220 订阅

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Two digital video encoder circuits

Two digital video encoder circuits

引用

advanced signal processing algorithms, architectures, and implementations III

作者： Eldon, John A. TRW LSI Products Inc. La Jolla CA USA

ISBN: (纸本)081940943X

Central to `multimedia' image processing is the desire to encode computer graphics data into a standard television signal, complete with line, field, and color subcarrier synchronizing information. The numerous incompatibilities between television and computer display standards render this operation far less trivial than it sounds to anyone who hasn't worked with both types of signals. To simplify the task of encoding computer graphics signals into standard NTSC (North America and Japan) or PAL (most of Europe) television format for display, broadcast, or recording, TRW LSI Products Inc. has introduced the two newest members of it multimedia integrated circuit family, the TMC22090 and TMC22190 digital video encoders.

关键词： Video signal processing

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Progressive image decoding based on the inverse discrete cosine transform

Progressive image decoding based on the inverse discrete cos...

引用

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

作者： Winograd, JM Nawab, SH Boston Univ. (United States)

ISBN: (纸本)0819425842

A new architecture is presented for incremental computation of the two-dimensional inverse discrete cosine transform (2D-IDCT) in the context of progressive image decoding. This architecture offers advantages over existing DCT inversion techniques for systems used in conjunction with progressive image coding schemes or those that must operate in environments with severe and/or time-varying resource constraints (e.g. real-time and low-power systems). The use of a bit-serial distributed arithmetic approach comprised of parallel input-pruned 2D-IDCT processing elements enables low-quality image reconstructions to be quickly and efficiently obtained using only a subset of the DCT coefficient bit stream. Initial approximate reconstructions can be improved in subsequent stages of incremental refinement according to the availability of processing resources or DCT coefficient data. Analysis is presented of image degradation at successive stages, illustrating uniform improvement according to quantitative and perceptual criteria.

关键词： progressive image decoding discrete cosine transform incremental refinement approximate processing

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Efficient implementation of a projection-based wave front sensor

Efficient implementation of a projection-based wave front se...

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Conference on advanced signal processing algorithms, architectures and implementations XII

作者： Holder, J Cain, S Mantica, P Indiana Univ Purdue Univ ITT Aerosp Commun Div Ft Wayne IN 46805 USA

ISBN: (纸本)0819445584

In this paper, a new wave front sensor design that utilizes the benefits of image projections is described and analyzed. The projection-based wave front sensor is similar to a Shack-Hartman type wave front sensor, but uses a correlation algorithm as opposed to a centroiding algorithm to estimate optical tilt. This allows the projection-based wave front sensor to estimate optical tilt parameters while guiding off of point sources and extended objects at very low signal to noise ratios. The implementation of the projection-based wave front sensor is described in detail showings important signal processing steps on and off of the focal plane array of the sensor. In this paper the design is tested in simulation for speed and accuracy by processing simulated astro-nomical data. These simulations demonstrate the accuracy of the projection-based wave front sensor and its superior performance to that of the traditional Shack-Hartman wave front sensor. Timing analysis is presented which shows how the collection and processing of image projections is computationally efficient and lends itself to a wave front sensor design that can produce adaptive optical control signals at speeds of up to 500 hz.

关键词： Wavefront sensors Charge-coupled devices signal to noise ratio Image sensors Sensors Adaptive optics signal processing Detection and tracking algorithms Image processing Astronomy

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Time sequenced adaptive filtering using a modified P-vector algorithm 6

Time sequenced adaptive filtering using a modified P-vector ...

引用

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

作者： Williams, R USAF WRIGHT LABAUTOMAT TARGET RECOGNIT DEV BRANCHWRIGHT PATTERSON AFBOH 45433

ISBN: (纸本)0819422347

An adaptive algorithm and two stage filter structure were developed for adaptive filtering of certain classes of signals that exhibit cyclostationary characteristics. The new modified P-vector algorithm (mPa) eliminates the need for a separate desired signal which is typically required by conventional adaptive algorithms. It is then implemented in a time-sequenced manner to counteract the nonstationary characteristics typically found in certain radar and bioelectromagnetic signals. Initial algorithm testing is performed on evoked responses generated by the visual cortex of the human brain with the objective, ultimately, to transition the results to radar signals. Each sample of the evoked response is modeled as the sum of three uncorrelated signal components, a time-varying mean (M), a noise component (N), and a random jitter component (Q). A two stage single channel time-sequenced adaptive filter structure was developed which improves convergence characteristics by de coupling the time-varying mean component from the `Q' and noise components in the first stage. The EEG statistics must be known a priori and are adaptively estimated from the pre stimulus data. The performance of the two stage mPa time-sequenced adaptive filter approaches the performance for the ideal case of an adaptive filter having a noiseless desired response.

关键词： time sequenced adaptive filter EEG biomedical signal processing radar signal processing

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MULTIsignal PROCESSOR SYSTEMS FOR ACOUSTICAL ECHO CONTROL

MULTISIGNAL PROCESSOR SYSTEMS FOR ACOUSTICAL ECHO CONTROL

引用

1990 INTERNATIONAL SYMP ON CIRCUITS AND SYSTEMS ( ISCAS 90 )

作者： FLEISCHMANN, G FRENZEL, R HATTY, B BJARNASON, E Institut fuer Netzwerk- und Signaltheorie Tech Hochschule Darmstadt West Germany

Two multi-digital-signal-processor systems for high-speed signal processing are presented. Both systems can be configured with a variable number of processor units. Thus, there is sufficient computational power for real-time implementations of high-order adaptive filter algorithms as required for echo cancellation, e.g., in hands-free telephone equipment. One system is based on the DSP32 floating-point DSP (digital signal processor) from AT&T. Monitoring and debugging is done with a standard PC system. The second system uses Fujitsu's MB8764 fixed-point DSP. It is controlled by an 80186 AMS-M-Bus microcomputer board. The architectures are designed to support efficient parallel implementations of high-order adaptive algorithms and of multirate systems for echo compensation in frequency subbands (i.e., block processing).

关键词： signal processing

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Operational data processing to improve SPOT image resolution

Operational data processing to improve SPOT image resolution

引用

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

作者： Breton, E Latry, C Lévy, F Rougé, B IGN F-31055 Toulouse France

ISBN: (纸本)0819441880

Mainly digital images are under sampled. It is the same for SPOT digital image satellite. The very meaning is that the instrument is too much powerful for this sampling. The worth side effect is that artifacts (aliasing) are introduced in the image, the good side is that images can be improved if the sampling density is increased. In this paper we use images from the two HRVIR instruments onboard SPOT1-4 satellite to multiply by a factor two the density and the resolution of the image.

关键词： super resolution operational data processing interlaced sampling interpolation correlation SPOT

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A higher order method for concentrating the STFT

A higher order method for concentrating the STFT

引用

advanced signal processing algorithms, architectures, and implementations XV

作者： Nelson, D.J. Smith, D.C. R61 U.S. Dept. of Defense Ft. Meade MD 20755 United States

We present a method for computing the theoretically exact estimate of the instantaneous frequency of a signal from local values of its short time Fourier transform under the assumption that the complex logarithm of the signal is a polynomial in time. We apply the method to the problem of estimating and separating non-stationary components of a multi-component signal. signal estimation and separation is based on a linear TF model in which the value of the signal at each time is distributed in frequency. This is a significant departure from the conventional nonlinear model in which signal energy is distributed in time and frequency. We further demonstrate by a simple example that IF estimated by the higher order method is significantly better than previously used first order methods.

关键词： signal processing

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Wavelet windowed time-frequency distribution decompositions

Wavelet windowed time-frequency distribution decompositions

引用

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

作者： Williams, WJ Sang, TH ONeill, JC Zalubas, EJ Univ. of Michigan (United States)

ISBN: (纸本)0819425842

This paper outlines means of combining and reconciling concepts associated with Cohen's class of distributions and with the wavelet transform. Both have their assets and their liabilities. Previous work has shown that one can decompose any time-frequency distribution (TFD) in Cohen's class into a weighted sum of spectrogram. A set of orthogonal analysis windows which also have the scaling property in common with wavelets is proposed. Successful application of this theory offers very fast computation of TFDs, since very few analysis windows may be needed and fast algorithms can be used. In addition, the decomposition idea offers the possibility of shaping the analysis such that good local and global properties as well as a number of desirable TFD properties are retained. Finally, one may view the result in terms of conventional Cohen's class concepts or, alternatively, in terms of wavelet concepts and potentially combine powerful insights and concepts from both points of view. Preliminary results applied to radar backscatter are provided. Performance curves for several wavelet types are also provided.

关键词： time-frequency wavelets signal decomposition radar fast algorithms

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Example of combined algorithm development and architecture design (Keynote Paper)

Example of combined algorithm development and architecture d...

引用

advanced signal processing algorithms, architectures, and implementations III

作者： Deprettere, Ed F. Delft Univ. of Technology Delft Netherlands

Matrix computation and signal processing applications can be given effective and efficient concrete forms in a systematic way if (1) the originating problems have formal and complete specifications and (2) the associa... 详细信息

ISBN: (纸本)081940943X

关键词： signal processing

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Convergence results on adaptive approximate filtering 6

Convergence results on adaptive approximate filtering

引用

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

作者： Ludwig, JT Nawab, SH Chandrakasan, AP MIT ELECT RES LABCAMBRIDGEMA 02139

ISBN: (纸本)0819422347

We present an IIR filtering technique based upon a recently proposed approach for reducing power consumption in implementation of frequency-selective filters. The basic idea in such techniques is to utilize the most recent input and output signal samples to estimate the current SNR (defined as the ratio of the in-band signal power to the out-of-band signal power) at the filter's input. This estimated input SNR is then used to update the filter order to the minimum value which would guarantee a minimum tolerable SNR at the filter's output. A key issue addressed in this paper is how well the estimated filter order converges to the theoretical minimum for situations satisfying the assumptions behind the derivation of the technique. Experimental results are used to verify that convergence to the correct filter order depends (1) upon the number of input and output samples used for estimating the input SNR, (2) upon the filter order applied in generating the output samples that are used in estimating the input SNR and (3) upon the proximity of the actual input SNR to boundaries in the input-SNR space corresponding to changes in the optimal choice for filter order.

关键词： low power approximate processing incremental refinement structure frequency-selective filtering Butterworth filters adaptive filtering

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