{"title":"Usage of smart antenna for cancelling neighboring base-station interferences in wireless CDMA communications","authors":"W. Ye, Y. Bar-Ness, A. Haimovich","doi":"10.1109/ACSSC.1997.680523","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680523","url":null,"abstract":"The capacity of wireless CDMA systems in the forward link is limited by both intra-cell and inter-cell cochannel interferences. In particular, when the mobile is close to a cell boundary, the desired signal from home base station (BS) is disturbed by relatively strong interference from neighboring BSs. In this paper, a receiver architecture is suggested at the mobile to utilize a small two-antenna array for interference cancellation. Such a canceller works well only when the channel vector of desired signal is known. We use the identifying spreading codes (as in IS-95 for example) to provide an adaptive channel vector estimate, and control the beam steering weight, hence improving the system capacity.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"26 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":"130497113","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}
{"title":"PCLS optimization of complex FIR digital filters and windows","authors":"J. L. Sullivan, J.W. Adams","doi":"10.1109/ACSSC.1997.680533","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680533","url":null,"abstract":"Peak-constrained least-squares (PCLS) optimization is a relatively new approach to solving digital filter design problems. Motivations for the PCLS optimality criterion are discussed previously. However, the PCLS examples and design methods proposed by Adams (1991) and Adams, Sullivan and Hashemi (1993) are focused on real symmetric FIR digital filters and windows. In this paper we extend the PCLS approach to the design of complex FIR digital filters and windows.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"78 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":"128598152","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}
{"title":"Adaptive detection of polynomial-phase signals embedded in noise using high-order ambiguity functions","authors":"S. Barbarossa, R. Mameli, A. Scaglione","doi":"10.1109/ACSSC.1997.679100","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679100","url":null,"abstract":"The parameter estimation of polynomial-phase signals (PPS) has been extensively studied in the literature in view of possible important applications in remote sensing as well as in telecommunications. However, the detection of PPS has not received similar attention. We propose and analyze an adaptive method for the detection of PPS embedded in white Gaussian noise based on the use of the so called product high order ambiguity function.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"14 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":"134176378","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}
{"title":"Polyphase implementation of a video scalar","authors":"A. Ramaswamy, Y. Nijim, W. Mikhael","doi":"10.1109/ACSSC.1997.679190","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679190","url":null,"abstract":"Image scaling is important in the conversion between different formats such as NTSC, PAL, HDTV and between the CCIR 601 video resolution to the various sizes included in MPEG coding. The change of the resolution proves specially beneficial for improving the coding efficiency. The scaling operation can be generalized by decimation by a factor of M followed by filtering and then interpolation by a factor of L where M and L are integers. The choice of the digital filter depends on the values of L and M. For certain resolution changes, M and L can be rather large integers. Conventional implementation of the filter may result in huge memory and computational requirements. To reduce this factor for practical applications, a polyphase implementation of the digital filter for the video scalar is presented. Some factors determining the choice of the digital filter are discussed. Finally, examples are shown for different resolution scaling of an image.","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":"130307771","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}
{"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}
{"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}
{"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}
{"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}
{"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}
{"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}