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Proceedings of IEEE 6th Digital Signal Processing Workshop最新文献

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The Multimedia Video Processor (MVP): a chip architecture for advanced DSP applications 多媒体视频处理器(MVP):一种用于高级DSP应用的芯片架构
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379881
R. Gove
{"title":"The Multimedia Video Processor (MVP): a chip architecture for advanced DSP applications","authors":"R. Gove","doi":"10.1109/DSP.1994.379881","DOIUrl":"https://doi.org/10.1109/DSP.1994.379881","url":null,"abstract":"The Texas Instruments Multimedia Video Processor (MVP), a single-chip multi-processing DSP for performing a variety of functions, provides unprecedented computing power and programmability. The MVP is particularly well suited for image and video applications. Utilizing a flexible parallel architecture, the chip can satisfy many video, audio, imaging and graphics functions, such as those required for desktop multimedia.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"51 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131773441","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
Optimal initial phase wavelet transform 最优初始相位小波变换
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379844
F. Bao, N. Erdol
{"title":"Optimal initial phase wavelet transform","authors":"F. Bao, N. Erdol","doi":"10.1109/DSP.1994.379844","DOIUrl":"https://doi.org/10.1109/DSP.1994.379844","url":null,"abstract":"Localization of the wavelet series coefficients and the efficiency of representation with the wavelet transform depends on the relative initial phase of the signal and the analyzing wavelet functions, and the dependence becomes more pronounced for wideband signals. The authors a) establish the significance of initial phase for efficient representation of narrowband signals using wavelets, b) develop an algorithm of linear computational complexity to search through a uniformly tiled time-scale plane for the optimal initial phase.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129746363","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 5
Comparison of optimal recovery based FIR synthesis filters with truncated ideal solutions 基于截断理想溶液的最优恢复FIR合成滤波器的比较
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379840
S. Cabrera, Sing-Wai Wu, G. Gonzalez
{"title":"Comparison of optimal recovery based FIR synthesis filters with truncated ideal solutions","authors":"S. Cabrera, Sing-Wai Wu, G. Gonzalez","doi":"10.1109/DSP.1994.379840","DOIUrl":"https://doi.org/10.1109/DSP.1994.379840","url":null,"abstract":"The topic of multirate filter banks and their use in subband coding for one-dimensional signals and images remains a very active research area in signal processing. The authors investigate theory and procedures for the problem of obtaining optimal synthesis filters for the recovery of the input from a general multirate filterbank decomposition including nonmaximally-decimated cases. The goals and results of the paper include: to illustrate the nature of the problem; to indicate the unconstrained approaches based on z-transform domain matrix pseudoinverses which usually do not yield causal and stable reconstruction filters; to illustrate an optimal approach that gives constrained-length (CL)FIR filters; to compare the FIR approach to the pseudoinverse filters; and to illustrate the role of a regularization parameter in the solution for the CLFIR filters. The comparison indicates the superiority of the CLFIR reconstruction system over equal length truncated pseudoinverse filters. The authors also find for the CLFIR that increasing the regularization parameter leads to reduced sensitivity to additive errors on the subband samples. On the other hand, the system deviates more from perfect reconstruction as this parameter increases.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"80 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125394496","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
MA-model identification using modulated cumulants 使用调制累积量的ma模型识别
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379853
T. Kaiser
{"title":"MA-model identification using modulated cumulants","authors":"T. Kaiser","doi":"10.1109/DSP.1994.379853","DOIUrl":"https://doi.org/10.1109/DSP.1994.379853","url":null,"abstract":"In this paper we present a new linear method for estimating the parameters of a moving average model from modulated cumulants of the observations of the system output. The input sequence must be non-Gaussian with some special properties described in the text. Both recursive closed-form and batch least-squares versions of the parameter estimator are presented. The proposed linear method utilizes a complete set of the relevant output statistics, so it should lead to more accurate parameter estimates compared to other linear methods. This property is illustrated through simulations. Furthermore it uses two different cumulants of arbitrary order and is therefore not restricted to the second and third order case.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126358030","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
Time-varying filter banks and multiwavelets 时变滤波器组和多小波
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379836
Martin Vetterli, G. Strang
{"title":"Time-varying filter banks and multiwavelets","authors":"Martin Vetterli, G. Strang","doi":"10.1109/DSP.1994.379836","DOIUrl":"https://doi.org/10.1109/DSP.1994.379836","url":null,"abstract":"A wavelet construction by Geronimo, Hardin and Massopust uses more than one wavelet and scaling function. Strang and Strela gave a filter bank interpretation of that result, as well as a condition for moment properties of the resulting wavelets. The present authors are concerned with the regularity of the resulting iterated filter bank scheme, that is, a matrix extension of the classic result by Daubechies (1988) on iterated filters. They show in particular: (i) the relation between time-varying filter banks and multiwavelets, (ii) the construction of multiwavelets as limits of iterated time-varying filter banks, (iii) a necessary condition for the convergence of the iterated matrix product and (iv) an exploration of examples of multiwavelets as iterations of time-varying filter banks.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126790495","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 41
Preprocessing for noise reduction of chaotic signals 混沌信号的降噪预处理
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379871
Chungyong Lee, D. Williams
{"title":"Preprocessing for noise reduction of chaotic signals","authors":"Chungyong Lee, D. Williams","doi":"10.1109/DSP.1994.379871","DOIUrl":"https://doi.org/10.1109/DSP.1994.379871","url":null,"abstract":"Existing algorithms for filtering noise from contaminated chaotic signals require relatively high signal-to-noise ratios (SNRs). A preprocessing method is proposed for initial noise reduction in severe SNRs. This method uses all the difference vectors between the points in a small neighborhood of the point of interest to estimate more accurately the tangent surface described by Cawley and Hsu (1992). This preprocessing method is applicable to noisy data from any nonlinear dynamical system, because it does not require knowledge of the system dynamics.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128841119","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Sparse deconvolution using Gaussian mixtures 使用高斯混合稀疏反褶积
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379866
I. Santamaria-Caballero, A. Figueiras-Vidal
{"title":"Sparse deconvolution using Gaussian mixtures","authors":"I. Santamaria-Caballero, A. Figueiras-Vidal","doi":"10.1109/DSP.1994.379866","DOIUrl":"https://doi.org/10.1109/DSP.1994.379866","url":null,"abstract":"We present a new algorithm to recover a sparse signal from a noisy register. The algorithm assumes a new model for the sparse signal that consists of a mixture of narrowband and broadband Gaussian noise both with zero mean. A penalty term which favors solutions driven from this model is added to the usual error cost function and the resultant global cost function is minimized with a gradient-type algorithm. We propose methods for updating the mixture parameters as well as for choosing the weighting parameter for the penalty term. Simulation experiments show that the accuracy of the proposed method is competitive with classical statistical detectors with a lower computational load. The proposed algorithm shows also a good performance when applied to a practical seismic deconvolution problem.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127172379","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
Wavelet representations for time-frequency concentrated signals 时频集中信号的小波表示
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379845
Jie Liang, T. Parks
{"title":"Wavelet representations for time-frequency concentrated signals","authors":"Jie Liang, T. Parks","doi":"10.1109/DSP.1994.379845","DOIUrl":"https://doi.org/10.1109/DSP.1994.379845","url":null,"abstract":"Time-frequency concentrated signals are defined in the paper as the class of signals whose Wigner distributions are concentrated in some region of the Wigner domain. The authors introduce the concept of the Kolmogorov n-width and the constrained n-width to quantitatively measure the ability of a basis to represent a time-frequency concentrated signal class (the cone-class signals). They select the best wavelet representation by comparing the constrained n-widths of different wavelet bases. An explicit formula is given to compute the Kolmogorov n-width for the cone-class signals. A globally convergent algorithm is proposed to calculate the constrained n-width for a given basis.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116768876","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Separation of co-channel FM/PM signals using the discrete polynomial-phase transform 用离散多项式相位变换分离同信道调频/调频信号
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-10-02 DOI: 10.1109/DSP.1994.379887
Meherwan Polad, Benjamin Friedlmder
{"title":"Separation of co-channel FM/PM signals using the discrete polynomial-phase transform","authors":"Meherwan Polad, Benjamin Friedlmder","doi":"10.1109/DSP.1994.379887","DOIUrl":"https://doi.org/10.1109/DSP.1994.379887","url":null,"abstract":"We consider the problem of separating two constant-amplitude continuous-phase signal components overlapping in time and frequency band. The discrete polynomial-phase transform (DPT) is used to estimate the parameters of the phase function of each component. Simulations are used to evaluate the performance of the algorithm.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-10-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114957970","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 9
Extraction of the inphase and quadrature components from oversampled bandpass signals using multistage decimator with BPFs and its performance evaluation 利用带bp滤波器的多级抽取器提取过采样带通信号的相位分量和正交分量及其性能评价
Proceedings of IEEE 6th Digital Signal Processing Workshop Pub Date : 1994-09-25 DOI: 10.1109/DSP.1994.379885
T. Sekiguchi, T. Kirimoto
{"title":"Extraction of the inphase and quadrature components from oversampled bandpass signals using multistage decimator with BPFs and its performance evaluation","authors":"T. Sekiguchi, T. Kirimoto","doi":"10.1109/DSP.1994.379885","DOIUrl":"https://doi.org/10.1109/DSP.1994.379885","url":null,"abstract":"We present a method of extracting the digital inphase (I) and quadrature (Q) components from oversampled bandpass signals using narrow-band bandpass Hilbert transformers. Down-conversion of the digitized IF signals to baseband and reduction of the quantization noise are accomplished by the multistage decimator with the complex coefficient bandpass digital filters (BPFs), of which the bandpass Hilbert transformers are composed. Most of the complex coefficient BPFs in the multistage decimator can be replaced with the lowpass filters (LPFs) under some conditions, which reduces computational burden. We evaluate the signal to quantization noise ratio of the I and Q components for the sinusoidal input by computer simulation. Simulation results show that the equivalent amplitude resolution of the I and Q components can be increased by 3 bits in comparison with non-oversampling case.<<ETX>>","PeriodicalId":189083,"journal":{"name":"Proceedings of IEEE 6th Digital Signal Processing Workshop","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-09-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125662156","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
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