Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.680250
Bo Tao, M. Orchard
This paper presents an analysis of the block-decimated motion estimates and relates them to the underlying motion random field. It further parameterizes the scene intensity random field and the motion random field in terms of their correlation properties. Within this framework, we develop an algorithm to optimize the window for overlapped block motion compensation as a function of the model parameters. Through simulations, we demonstrate that the optimal window resulting from the parametric formulation offers performance comparable to the window deterministically optimized for the test sequence, and it offers more robust performance outside the training set. Finally, we apply our algorithm to adapt the overlapped window to match the temporally changing characteristics of the scene and motion fields. We demonstrate that for real-time applications, where the number of frames used for adapting the window is limited, our algorithm significantly outperforms the method introduced by Orchard and Sullivan (see IEEE Trans. Image Processing, vol.3, no.5, p.693-9, 1994).
{"title":"Window design for overlapped block motion compensation through statistical motion modeling","authors":"Bo Tao, M. Orchard","doi":"10.1109/ACSSC.1997.680250","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680250","url":null,"abstract":"This paper presents an analysis of the block-decimated motion estimates and relates them to the underlying motion random field. It further parameterizes the scene intensity random field and the motion random field in terms of their correlation properties. Within this framework, we develop an algorithm to optimize the window for overlapped block motion compensation as a function of the model parameters. Through simulations, we demonstrate that the optimal window resulting from the parametric formulation offers performance comparable to the window deterministically optimized for the test sequence, and it offers more robust performance outside the training set. Finally, we apply our algorithm to adapt the overlapped window to match the temporally changing characteristics of the scene and motion fields. We demonstrate that for real-time applications, where the number of frames used for adapting the window is limited, our algorithm significantly outperforms the method introduced by Orchard and Sullivan (see IEEE Trans. Image Processing, vol.3, no.5, p.693-9, 1994).","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"160 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122864142","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.679107
K. Johnson
Constructing multiwavelet-based filter banks (multifilters) is more difficult than constructing scalar wavelet filters, partly because the noncommutativity of matrix multiplication prevents a trivial extension of the scalar wavelet "flip construction". Commutative multifilters avoid this problem by allowing the flip construction to be used, thereby simplifying the design process. This paper presents a condition on the polyphase components of the analysis multifilters which is necessary and sufficient for the multifilters to achieve commutativity in addition to perfect reconstruction and orthogonality. This condition involves matrices of a form similar to that appearing in various other contexts, some of which are discussed. The paper includes simple examples of multifilters satisfying the condition obtained.
{"title":"A necessary and sufficient condition for commutative PR orthogonal multifilter banks","authors":"K. Johnson","doi":"10.1109/ACSSC.1997.679107","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679107","url":null,"abstract":"Constructing multiwavelet-based filter banks (multifilters) is more difficult than constructing scalar wavelet filters, partly because the noncommutativity of matrix multiplication prevents a trivial extension of the scalar wavelet \"flip construction\". Commutative multifilters avoid this problem by allowing the flip construction to be used, thereby simplifying the design process. This paper presents a condition on the polyphase components of the analysis multifilters which is necessary and sufficient for the multifilters to achieve commutativity in addition to perfect reconstruction and orthogonality. This condition involves matrices of a form similar to that appearing in various other contexts, some of which are discussed. The paper includes simple examples of multifilters satisfying the condition obtained.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"185 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126784323","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.679183
C. S. Agate, R. Iltis
The statistics of the reduced sufficient statistics (RSS) estimator are derived for the nonlinear additive white Gaussian noise measurement model. The RSS algorithm recursively propagates a set of sufficient statistics for a mixture density which approximates the true posterior density of a parameter vector. The joint probability density function for the weighting coefficients of the mixture density is derived for the case of additive white Gaussian noise. Through integration of this density, the estimator bias and mean-squared error are determined. The results are applied to a scalar estimation problem in which the sample-averaged statistics are compared to those derived from numerical integration of the density function.
{"title":"Small sample properties of the RSS estimation algorithm for Gaussian measurement noise","authors":"C. S. Agate, R. Iltis","doi":"10.1109/ACSSC.1997.679183","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679183","url":null,"abstract":"The statistics of the reduced sufficient statistics (RSS) estimator are derived for the nonlinear additive white Gaussian noise measurement model. The RSS algorithm recursively propagates a set of sufficient statistics for a mixture density which approximates the true posterior density of a parameter vector. The joint probability density function for the weighting coefficients of the mixture density is derived for the case of additive white Gaussian noise. Through integration of this density, the estimator bias and mean-squared error are determined. The results are applied to a scalar estimation problem in which the sample-averaged statistics are compared to those derived from numerical integration of the density function.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126775925","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.680540
C. V. Schimpfle, P. Rieder, J. Nossek
A new efficient implementation of a IEEE-standard conform 8 point discrete cosine transform (DCT) is presented. The architecture is based on different classes of orthogonal 2/spl times/2 /spl mu/-rotations used to approximate the angles of the DCT. By using only orthogonal /spl mu/-rotations it is guaranteed, that the whole transform remains orthogonal and perfect reconstruction of the signal can be achieved. It is shown that for the implementation of the DCT with approximated rotation angles (angle quantization) about 28% less shift and add operations are necessary than for a standard conform implementation with coefficient quantization. This lends to a large power benefit due to less adder hardware and less capacitive load of the global interconnects. Besides this, there are some other advantageous aspects concerning the area and delay. To support the full custom design of the layout, module generators for all the different classes /spl mu/-rotations can be used to generate the necessary rotations automatically.
{"title":"A power efficient implementation of the discrete cosine transform","authors":"C. V. Schimpfle, P. Rieder, J. Nossek","doi":"10.1109/ACSSC.1997.680540","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680540","url":null,"abstract":"A new efficient implementation of a IEEE-standard conform 8 point discrete cosine transform (DCT) is presented. The architecture is based on different classes of orthogonal 2/spl times/2 /spl mu/-rotations used to approximate the angles of the DCT. By using only orthogonal /spl mu/-rotations it is guaranteed, that the whole transform remains orthogonal and perfect reconstruction of the signal can be achieved. It is shown that for the implementation of the DCT with approximated rotation angles (angle quantization) about 28% less shift and add operations are necessary than for a standard conform implementation with coefficient quantization. This lends to a large power benefit due to less adder hardware and less capacitive load of the global interconnects. Besides this, there are some other advantageous aspects concerning the area and delay. To support the full custom design of the layout, module generators for all the different classes /spl mu/-rotations can be used to generate the necessary rotations automatically.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"71 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130203297","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.679072
K. Anandakumar, S. Kassam
We consider wavelet-based self-affine modeling of speech signals for speech compression. We propose two approaches. In the first approach, the self-affine modeling is considered for the representation of speech signal itself. In the second approach, the self-affine modeling is applied for the representation of speech excitation of a linear predictor. In both approaches, error propagation at reconstruction due to the modeling error is avoided by using a causal domain pool. We compare the performance of proposed schemes with that of the GSM 06.10 standard.
{"title":"Self-affine modeling of speech signal in speech compression","authors":"K. Anandakumar, S. Kassam","doi":"10.1109/ACSSC.1997.679072","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679072","url":null,"abstract":"We consider wavelet-based self-affine modeling of speech signals for speech compression. We propose two approaches. In the first approach, the self-affine modeling is considered for the representation of speech signal itself. In the second approach, the self-affine modeling is applied for the representation of speech excitation of a linear predictor. In both approaches, error propagation at reconstruction due to the modeling error is avoided by using a causal domain pool. We compare the performance of proposed schemes with that of the GSM 06.10 standard.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"226 ","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"113983487","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.680537
M. Lutovac, D. Tosic, B. Evans
Classical filter design techniques return only one design from an infinite collection of alternative designs, or fail to design filters when solutions exist. These classical techniques hide a wealth of alternative filter designs that are more robust when implemented in analog circuits, digital hardware, and embedded software. We present (1) case studies of optimal analog and digital IIR filters that cannot be designed with classical techniques, and (2) the formal, mathematical framework that underlies their solutions. We have automated the advanced filter design techniques in software.
{"title":"Advanced filter design","authors":"M. Lutovac, D. Tosic, B. Evans","doi":"10.1109/ACSSC.1997.680537","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680537","url":null,"abstract":"Classical filter design techniques return only one design from an infinite collection of alternative designs, or fail to design filters when solutions exist. These classical techniques hide a wealth of alternative filter designs that are more robust when implemented in analog circuits, digital hardware, and embedded software. We present (1) case studies of optimal analog and digital IIR filters that cannot be designed with classical techniques, and (2) the formal, mathematical framework that underlies their solutions. We have automated the advanced filter design techniques in software.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124078584","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.679080
H. Card
In this paper the properties of artificial neural network computations by digital VLSI systems are discussed. We also comment on artificial computational models, learning algorithms, and digital implementations of ANNs in general. The analysis applies to regular arrays or processing elements performing binary integer arithmetic at various bit precisions. Computation rates are limited by power dissipation which is dependent upon required precision and packaging constraints such as pinout. They also depend strongly on the minimum feature size of the CMOS technology. We emphasize custom digital implementations with low bit precision, because these circuits require reduced power and silicon area. One way this may be achieved is using stochastic arithmetic, with pseudorandom number generation based on cellular automata circuits.
{"title":"Limits to neural computations in digital arrays","authors":"H. Card","doi":"10.1109/ACSSC.1997.679080","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679080","url":null,"abstract":"In this paper the properties of artificial neural network computations by digital VLSI systems are discussed. We also comment on artificial computational models, learning algorithms, and digital implementations of ANNs in general. The analysis applies to regular arrays or processing elements performing binary integer arithmetic at various bit precisions. Computation rates are limited by power dissipation which is dependent upon required precision and packaging constraints such as pinout. They also depend strongly on the minimum feature size of the CMOS technology. We emphasize custom digital implementations with low bit precision, because these circuits require reduced power and silicon area. One way this may be achieved is using stochastic arithmetic, with pseudorandom number generation based on cellular automata circuits.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127768122","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.680023
M. L. Kramer, D.L. Jones
Transform-based interference suppression works well for stationary signals, but when applied to nonstationary interference, it is often beneficial to use a larger overdetermined library of transforms. We propose a scheme which uses the same project/threshold technique used in traditional transform-based excision applied to an overdetermined frame. Such a frame can consist of a variety of elements, including time and frequency invariant sets of wavelets and chirps, to help characterize a larger variety of interferers. For thresholds within certain bounds, the proposed method guarantees improvement in the effective signal-to-interference ratio (SIR). Analytical and simulation results illustrate the effectiveness of the technique for a variety of interference types, including combinations of non-orthogonal interferers. The study is of interest in DS-SS communications.
{"title":"Nonstationary interference suppression using adaptive overdetermined frame representations","authors":"M. L. Kramer, D.L. Jones","doi":"10.1109/ACSSC.1997.680023","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680023","url":null,"abstract":"Transform-based interference suppression works well for stationary signals, but when applied to nonstationary interference, it is often beneficial to use a larger overdetermined library of transforms. We propose a scheme which uses the same project/threshold technique used in traditional transform-based excision applied to an overdetermined frame. Such a frame can consist of a variety of elements, including time and frequency invariant sets of wavelets and chirps, to help characterize a larger variety of interferers. For thresholds within certain bounds, the proposed method guarantees improvement in the effective signal-to-interference ratio (SIR). Analytical and simulation results illustrate the effectiveness of the technique for a variety of interference types, including combinations of non-orthogonal interferers. The study is of interest in DS-SS communications.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"231 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121409020","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.679123
I. Kontoyiannis
We present an overview of several recent results (some new and some known) on the asymptotic performance of different variants of the Lempel-Ziv coding algorithm, in both the lossless case and the lossy case. The results are based on the asymptotic behavior of waiting times, following the general methodology introduced by Wyner and Ziv (1989). We show that, in this framework, very precise statements can be made about the second-order (asymptotic) properties of the codeword lengths.
{"title":"Second-order analysis of lossless and lossy versions of Lempel-Ziv codes","authors":"I. Kontoyiannis","doi":"10.1109/ACSSC.1997.679123","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679123","url":null,"abstract":"We present an overview of several recent results (some new and some known) on the asymptotic performance of different variants of the Lempel-Ziv coding algorithm, in both the lossless case and the lossy case. The results are based on the asymptotic behavior of waiting times, following the general methodology introduced by Wyner and Ziv (1989). We show that, in this framework, very precise statements can be made about the second-order (asymptotic) properties of the codeword lengths.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115899540","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}
Pub Date : 1997-11-02DOI: 10.1109/ACSSC.1997.680048
N. Magotra, Yannuo Yang, R. Whitman, P. Kasthuri
This paper presents research results involving the development of a speech enhancement algorithm for wireless communication systems. The algorithm has also been designed to be amenable to real-time processing on Texas Instrument's TMS320C3X digital signal processing (DSP) chip. The algorithm uses adaptive single-channel correlation, or higher order cumulants, to achieve its goal of noise reduction. The algorithm has been tested using the CTIMIT data base. The paper presents some quantitative results and outlines some subjective assessments that we intend to perform in our continuation of this research.
{"title":"Real time speech enhancement for wireless communication systems","authors":"N. Magotra, Yannuo Yang, R. Whitman, P. Kasthuri","doi":"10.1109/ACSSC.1997.680048","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680048","url":null,"abstract":"This paper presents research results involving the development of a speech enhancement algorithm for wireless communication systems. The algorithm has also been designed to be amenable to real-time processing on Texas Instrument's TMS320C3X digital signal processing (DSP) chip. The algorithm uses adaptive single-channel correlation, or higher order cumulants, to achieve its goal of noise reduction. The algorithm has been tested using the CTIMIT data base. The paper presents some quantitative results and outlines some subjective assessments that we intend to perform in our continuation of this research.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115988575","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}
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