{"title":"Fractional wave packet transform","authors":"Ying Huang, B. Suter","doi":"10.1109/DSPWS.1996.555549","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555549","url":null,"abstract":"The short-time Fourier transform (STFT), or windowed Fourier transform, is the most widely used method in signal processing. We introduce the concept of the fractional wave packet transform (FRWPT), based on the idea of the fractional Fourier transform (FRFT) and wave packet transform (WPT). We show a version of the resolution of the identity and some properties of FRWPT connected with those of the FRFT and WPT.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114292607","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":"Hybrid higher order cepstrum and functional link network (HOCFLN) based blind equaliser","authors":"A. Alkulaibi, J. Soraghan, A. Hussain","doi":"10.1109/DSPWS.1996.555558","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555558","url":null,"abstract":"A new hybrid higher order cepstrum (HOC) and functional link network (FLN) based blind equaliser (HOCFLN) is presented. The system initially uses the complex cepstrum of the 1-D slice of the fourth order cumulants of the unknown received signal to partially estimate the equaliser coefficients, then it switches to an FLN adaptive equaliser operating in the decision directed mode (DDM) to further improve the mean squared error (MSE) convergence. This method is flexible to accommodate both non-minimum phase MA and ARMA channels. It is shown that the new HOCFLN system gives significant performance improvements with less computational complexity compared to the conventional equalization algorithms. Performance results for channels exhibiting abrupt channel characteristic changes are also shown.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"276 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122854601","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":"Fractal and linear pyramids","authors":"J. Prades-Nebot, A. Albiol","doi":"10.1109/DSPWS.1996.555464","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555464","url":null,"abstract":"Pyramids are data structures with multiresolution information that have been applied successfully on many image processing and analysis tasks. We compare the properties of fractal and linear pyramids. The relations between the levels of a fractal pyramid are studied, generalising the results obtained by Baharav et al. (see Fractal Compression: Theory and applications to Digital Images, chapter 5, p.91-117. Springer-Verlag, New-York, 1995). As with linear pyramids, in order to go up one level in a fractal pyramid (decreasing resolution), a process of linear filtering and decimation must be iterated. We show that there is a direct relation between contraction and filter coefficients. Pyramids generated with several coefficient choices are also studied. The self-similarity property of PIFS (partitioned iterated function systems) becomes clear when descending one level in the fractal pyramid (increasing resolution), and unlike the case of linear pyramids, no detail signal must be added, because it is automatically created by the PIFS code.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131153503","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":"Properties of optimal, compaction filters in subband coding","authors":"P. Vaidyanathan","doi":"10.1109/DSPWS.1996.555467","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555467","url":null,"abstract":"Optimum compaction filters find application in the design of orthonormal subband coders with maximum coding gain. The compaction gain for a process is defined, and a number of its properties studied. Bounds on the compaction gain are established, and the conditions for the attainment of these bounds are specified in terms of the input power spectrum. Finally, the behavior of the optimum compaction gain as a function of M (the number of channels in a subband coder) is also addressed.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131378189","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":"Correlation performance of chaotic signals in spread spectrum systems","authors":"S.S. Rao, S. Howard","doi":"10.1109/DSPWS.1996.555573","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555573","url":null,"abstract":"The performance of chaotic sequences used in direct sequence spread spectrum (DS/SS) communications is assessed. It is demonstrated that this method leads to an improvement in signal detection performance over traditional spreading techniques. Other advantages of using chaotic dynamical systems for such applications are presented.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133578434","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":"Reconstruction in tomography from severe incomplete projection data using multiresolution analysis and optimization","authors":"J. Feng","doi":"10.1109/DSPWS.1996.555478","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555478","url":null,"abstract":"This paper proposed a new optimization approach over whole distribution region in tomography for smooth distributions from severe incomplete projection data. The multiresolution analysis proposed by Mallet (1989) based on wavelet transform with orthonormal bases proposed by Daubechies (1988) is applied to reduce the number of parameters for optimization and to ignore the details of wavelet representations at high resolution level. Results of reconstructing from 3 or 6 projections for two test distributions demonstrates the usefulness of this approach of tomography for severe incomplete projection data.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133068492","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":"Multirate modeling of human ear frequency resolution for hearing aids","authors":"M. Li, H. McAllister, N. Black","doi":"10.1109/DSPWS.1996.555484","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555484","url":null,"abstract":"A real-time multirate filtering design project, as an application in digital hearing aids, has been developed. The motivation is that it allows nonuniform frequency resolution which automatically leads us to use different decimators for different frequency bandwidths. Therefore, a variation of the tree structured multirate multistage filter bank is exploited with six pairs of polyphase filters for decimation and six pairs for interpolation to achieve seven different frequency bandwidths, each for an octave. Such nonuniform systems are well-suited for the processing of sound signals, because of the decreasing resolution of the human ear at higher frequencies. For audiogram match, the frequency response of the decimation filter bank can be individually adjustable in each octave band. The other advantages of multirate processing provide for less computational and storage requirements as well as lower order filter design. Texas Instruments Evaluation Module (EVM), based on the fifth generation TI TMS320C50 digital signal processor, is used for implementation.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"32 4","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114042256","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":"Symmetric adaptive predictive structure for tracking channel non-stationarities","authors":"S. B. Jebara, M. Jaidane-Saidane","doi":"10.1109/DSPWS.1996.555531","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555531","url":null,"abstract":"The basic idea of this paper is related to the fact that the steady state property in a non-stationary system context is strongly related to the input correlation characteristics. When the LMS algorithm is used to identify a system with variations modeled by a random walk, the performance is degradated as the input correlation increases. The classical identification scheme can be improved by the use of a prewhitening adaptive filter. A theoretical analysis of an adaptive predictive identification scheme is presented. This study illustrates the contribution of predictive structures for tracking system non-stationarities.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115814281","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":"A delayed error least mean squares adaptive filtering algorithm and its performance analysis","authors":"J. Thomas","doi":"10.1109/DSPWS.1996.555542","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555542","url":null,"abstract":"High sampling rate realizations of the least mean squares (LMS) adaptive filtering algorithm require that the inherent recursive computational bottleneck in the impulse response updating be broken by introducing algorithmic delays into the error feedback path. The well known delayed LMS (DLMS) technique achieves this by convolving delayed error samples with delayed input samples. This paper proposes a possible realization that convolves delayed error samples with undelayed input samples, motivated by systolization and pipelining requirements that use only the delays introduced in the error feedback path. We provide a convergence analysis of this delayed error LMS (DELMS) algorithm along with experimental simulations that prove the stability of this adaptation technique under desired operating conditions and improved tracking performance in nonstationary environments, compared with the DLMS algorithm.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124478902","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":"Noniterative reconstruction of band limited signals and coding techniques","authors":"J. Vieira, P. Ferreira","doi":"10.1109/DSPWS.1996.555482","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555482","url":null,"abstract":"In this paper we propose linear iterative and noniterative algorithms which can efficiently correct a subset of t samples of a signal, whose values might have been corrupted, possibly due to noise, clipping, or any other reasons. We discuss the applicability of techniques used in error correcting codes to this problem, and the possibility of determining the location of the erroneous samples in oversampled band limited signals. The method borrows ideas from error-correcting codes such as BCH codes, but works in the complex field rather than in a Galois field. This work complements works in reconstruction theory which usually assume that the positions of lost samples or pixels are known, and improves on previously reported nonlinear iterative algorithms.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123141786","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}