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 r...
详细信息
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.
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...
详细信息
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 signalprocessing 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.
We describe a method for the formal determination of signal bit width in fixed points VLSI implementations of signalprocessingalgorithms containing loop nests. The main contribution of this paper is the use of resul...
详细信息
ISBN:
(纸本)0769517129
We describe a method for the formal determination of signal bit width in fixed points VLSI implementations of signalprocessingalgorithms containing loop nests. The main contribution of this paper is the use of results of the (max, +) algebraic theory to find the integral bit width of algorithms containing loop nests whose bound parameters are not statically known. Combined with recent results on fractional bit width determination, this can be used for 1-dimensional systolic-like arrays implementing linear signalprocessingalgorithms. Although this technique is presented in the context of a specific high level design methodology (based on systems of affine recurrence equations), it can be used in many high level design environments.
One of the main goals of the STAP-BOY program has been the implementation of a, space-time adaptive processing (STAP) algorithm on graphics processing units (GPUs) with the goal of reducing the processing time. Within...
详细信息
ISBN:
(纸本)9780819472946
One of the main goals of the STAP-BOY program has been the implementation of a, space-time adaptive processing (STAP) algorithm on graphics processing units (GPUs) with the goal of reducing the processing time. Within the context of GPU implementation, we have further developed algorithms that exploit data redundancy inherent in particular STAP applications. Integration of these algorithms with GPU architecture is of primary importance for fast algorithmic processing times. STAP algorithms involve solving a linear system in which the transformation matrix is a covariance matrix. A standard method involves estimating a covariance matrix from a data matrix, computing its Cholesky factors by one of several methods. and then solving the system by substitution. Some STAP applications have redundancy in successive data matrices from which the covariance matrices are formed. For STAP applications in which a data matrix is updated with the addition of a new data row at the bottom and the elimination of the oldest data in the top of the matrix, a sequence of data matrices have multiple rows in common. Two methods have been developed for exploiting this type of data redundancy when computing Cholesky factors. These two methods are referred to as 1) Fast QR factorizations of successive data matrices 2) Fast Cholesky factorizations of successive covariance matrices. We have developed GPU implementations of these two methods. We show that these two algorithms exhibit reduced computational complexity when compared to benchmark algorithms that do not exploit data, redundancy. More importantly, we show that when these algorithmic improvements are optimized for the GPU architecture, the processing times of a GPU implementation of these matrix factorization algorithms may be greatly improved.
Blind source separation is an emerging field of fundamental research with a broad range of applications. It is motivated by practical problems that involve several source signals and several sensors. Each sensor recei...
详细信息
ISBN:
(纸本)0819425842
Blind source separation is an emerging field of fundamental research with a broad range of applications. It is motivated by practical problems that involve several source signals and several sensors. Each sensor receives an instantaneous linear mixture of the source signals. The problem of the blind source separation consists then of recovering the original waveforms of the sources without any knowledge of the mixture structure. So far, the problem of the blind source separation has been solved using statistical information available on the source signals. A blind source separation approach for non-stationary signals based on time-frequency representations (TFR) have been recently introduced by the authors (SPIE 1996). Herein, we generalize the TFR based blind source separation approach to arbitrary variables, including time and frequency. 'Spatial joint arbitrary variable distributions' are introduced and used for blind source separation via joint diagonalization techniques.
A robust focusing matrix, called unitary constrained array manifold focusing, is proposed for the coherent signal-subspace method which is a novel approach for direction finding of broad- band sources. The proposed fo...
详细信息
ISBN:
(纸本)081940943X
A robust focusing matrix, called unitary constrained array manifold focusing, is proposed for the coherent signal-subspace method which is a novel approach for direction finding of broad- band sources. The proposed focusing matrix has robustness against size variations of focusing sectors obtained by preliminary group angle estimates. To quantify information loss due to source focusing, a new measure, relative efficiency index, is introduced. This index is useful in assessing statistical sufficiency of a data-reduced statistic formed by source focusing. The sensitivity issue as well as estimation performance comparisons for various focusing schemes, based on simulations and relative efficiency index, are also presented.
Given two n. x n Toeplitz matrices T-1 and T-2, and a vector b epsilon R-n2, consider the linear system Ax = b - eta, where eta epsilon R-n2 is an unknown vector representing the noise and A = T-1 x T-2. Recovering ap...
详细信息
ISBN:
(纸本)0819437611
Given two n. x n Toeplitz matrices T-1 and T-2, and a vector b epsilon R-n2, consider the linear system Ax = b - eta, where eta epsilon R-n2 is an unknown vector representing the noise and A = T-1 x T-2. Recovering approximations of a, given A and b, is encountered in image restoration problems. We propose a method for the approximation of the solution a: that has good regularization properties. The algorithm is based on a modified version of Newton's iteration for matrix inversion and relies on the concept of approximate displacement rank. We provide a formal description of the regularization properties of Newton's iteration in terms of filters and determine bounds to the number of iterations that guarantee regularization. The method is extended to deal with more general systems where A = Sigma (h)(i=1) T-1((i)) x T-2((i)). The cost of computing regularized inverses is O(n log n) operations (ops), the cost of solving the system Ax = b is O(n(2) log n) ops. Numerical experiments which show the effectiveness of our algorithm are presented.
We investigate the estimation of both the number of waves and the wave parameters for transient wavefields in a geophysical application. Models of the earth have to be verified by seismogram analysis. We present two n...
详细信息
ISBN:
(纸本)0819412767
We investigate the estimation of both the number of waves and the wave parameters for transient wavefields in a geophysical application. Models of the earth have to be verified by seismogram analysis. We present two nonparametric methods for the estimation of wave parameters where one is based on the first-order stationary spectrum and the other one is based on the second-order cumulant spectrum. We also investigate a parametric method for wave parameter estimation which allows us to determine the number of incident waves. The algorithms are applied to synthetic seismic data.
We discuss the application of time-frequency analysis to biomechanical-type signals, and in particular to signals that would be encountered in the study of rotation rates of bicycle pedaling. We simulate a number of s...
详细信息
ISBN:
(纸本)9780819468451
We discuss the application of time-frequency analysis to biomechanical-type signals, and in particular to signals that would be encountered in the study of rotation rates of bicycle pedaling. We simulate a number of such signals and study how well they are represented by various time-frequency methods. We show that time-frequency representations track very well the instantaneous frequency even when there are very fast changes. In addition. we do a correlation analysis between time-series whose instantaneous frequency is changing and show that the traditional correlation coefficient is insufficient to characterize the correlations. We instead show that the correlation coefficient should be evaluated directly from the instantaneous frequencies of the time series, which can be easily estimated from their time-frequency distributions.
This paper discusses the use of a recently introduced index calculus Double-Base Number System (IDBNS) for representing and processing numbers for non-linear digital signalprocessing;the target application is a digit...
详细信息
ISBN:
(纸本)0819437611
This paper discusses the use of a recently introduced index calculus Double-Base Number System (IDBNS) for representing and processing numbers for non-linear digital signalprocessing;the target application is a digital hearing aid processor. The IDBNS representation uses 2 orthogonal bases (2 and 3) to represent real numbers with arbitrary precision. By restricting the number of digits to one or two, it is possible to efficiently represent the real number using the indices of the bases rather than the distribution of the digits. In this paper we discuss the use of the two-digit form of this representation (2-IDBNS) to efficiently perform arithmetic associated with the non-linear processing required to correct the usual forms of hearing loss in a digital hearing aid. The non-linear processing takes the form of dynamic range compression as a function of frequency band. Currently developed digital hearing instrument processors require large dynamic range representations (20-24 bits) in order to accurately generate the dynamic range compression associated with typical hearing loss. We show that the natural non-linear representation afforded by the IDBNS provides both a more efficient signal representation and a more efficient technique for processing the dynamic range compression. We pay particular attention to a novel technique of converting from a linear binary input directly to the 2-IDBNS representation using an observation of partial cyclic repetition in the indices along with near unity approximants.
暂无评论