In this paper we combine two recently developed multi-scale deconvolution algorithms, known as the scale-time domain method and the sum-of-cumulants domain method. We formulate the deconvolution problem in the scale-c...
详细信息
ISBN:
(纸本)0819437611
In this paper we combine two recently developed multi-scale deconvolution algorithms, known as the scale-time domain method and the sum-of-cumulants domain method. We formulate the deconvolution problem in the scale-cumulant domain using the Scale Transform (ST) and show that the procedure is simpler when the unknown source signal is non-minimum phase and robust if Gaussian noise exists.
Media signalprocessing requires high computing power and the algorithms exhibit a great deal of parallelism on low precision data. The basic components of multi-media objects are usually simple integers with 8, 12, o...
详细信息
ISBN:
(纸本)0819437611
Media signalprocessing requires high computing power and the algorithms exhibit a great deal of parallelism on low precision data. The basic components of multi-media objects are usually simple integers with 8, 12, or 16 bits of precision. In order to support efficient processing of media signals, Instructions Set Architecture (ISA) of the traditional processors requires modifications. In this paper, we present the quantitative analysis and the computational complexity required to perform media processing. Main classes of instructions that are needed for the required level of performance of the Media Processor are identified. Their efficient implementation and effect on the processor data-path is discussed. The main operations required in media processing are Addition (with or without saturation), Multiplication (with or without rounding), Sum of Products, and Average of two numbers.
We present a general procedure for obtaining equations of motion for the Wigner distribution of functions that are governed by ordinary and partial differential equations. For the case of fields we show that in genera...
详细信息
ISBN:
(纸本)0819437611
We present a general procedure for obtaining equations of motion for the Wigner distribution of functions that are governed by ordinary and partial differential equations. For the case of fields we show that in general one must consider Wigner distributions of the four variables, position, momentum, time and frequency. We also show that in general one cannot write an equation of motion for position and momentum however it can be done in some cases, the Schrodinger equation being one such case. Our method leads to an equation of motion for the Schrodinger equation with time dependent potentials in contrast to the result obtained by Wigner and Moyal which was for time independent potentials.
This study has been realized to improve industrial machines that allow to analyze planks by detecting their width and too important defects thanks to a computer vision system. These machines are currently piloted by s...
详细信息
ISBN:
(纸本)0819437611
This study has been realized to improve industrial machines that allow to analyze planks by detecting their width and too important defects thanks to a computer vision system. These machines are currently piloted by software with the help of PCs. The aim of our work is to realize a hardware card to increase the processing speed.
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 &qu...
详细信息
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.
The problem addressed in this paper is to removing unwanted sounds from music sound. The sound to be removed could be disturbance such as cough. We shall present some preliminary results on this problem using statisti...
详细信息
ISBN:
(纸本)0819437611
The problem addressed in this paper is to removing unwanted sounds from music sound. The sound to be removed could be disturbance such as cough. We shall present some preliminary results on this problem using statistical properties of signals. Our approach consists of three steps. We first estimate the fundamental frequencies and partials given noise-corrupted music sound. This gives us the autoregressive (AR) model of the music sound. Then se filter the noise-corrupted sound using the AR parameters. The filtered signal is then subtracted from the original noise-corrupted signal to get the disturbance. Finally the obtained disturbance is used as reference signal to eliminate the disturbance from the noise-corrupted music signal. Above three steps are carried out in a recursive manner using a sliding window or an infinitely growing window with an appropriate forgetting factor.
In this paper, we formulate a framework for discrete-time processing of Linear Scale Invariant (LSI) systems which are invariant to scale changes in time. Continuous-time LSI systems can be processed by Mellin and Mod...
详细信息
ISBN:
(纸本)0819437611
In this paper, we formulate a framework for discrete-time processing of Linear Scale Invariant (LSI) systems which are invariant to scale changes in time. Continuous-time LSI systems can be processed by Mellin and Modified Scale Transforms analogous to the use of the Laplace and Fourier Transforms in the continuous-time processing of LTI systems. In this work, we present the geometric sampling theorem to prevent aliasing in the scale domain. We also derive the perfect reconstruction filter in time domain, the discrete-time convolution sum, the Discrete Time Modified Scale Transform (DTMST) and the Discrete Modified Scale Transform (DMST) for geometrically sampled signals.
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.
Color representation and comparison based on the histogram has proved to be very efficient for image indexing in content-based image retrieval and machine vision applications. However, the issues of color constancy an...
详细信息
ISBN:
(纸本)0819437611
Color representation and comparison based on the histogram has proved to be very efficient for image indexing in content-based image retrieval and machine vision applications. However, the issues of color constancy and accurate color similarity measures remain unsolved. This paper presents a new algorithm for intensity-insensitive color characterization for image retrieval and machine vision applications. The color characterization algorithm divides the HSI (hue, saturation and intensity) color space into a given number of bins in such a way that the color characterization represents all the colors in the hue/saturation plane as well as black, white and gray colors. The color distribution in these bins of the HSI space is represented in the form of a one-dimensional vector called Color Spectrum Vector (CSV). The color information that is stored in the CSV is insensitive to changes in the luminance. A weighted version of CSV called WCSV is introduced to take the similarity of the neighboring bins into account A Fuzzy Color Spectrum Vector (FCSV) color representation vector that takes into account the human uncertainty in color classification process is also introduced here. The accuracy and speed of the algorithm is demonstrated in this paper through a series of experiments on image indexing and machine vision applications.
暂无评论