{"title":"Signal-to-noise Ratio Enhancement Of Cyclic Summation","authors":"D. McMahon, A. Bolton","doi":"10.1109/ISSPA.1996.615142","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615142","url":null,"abstract":"Many signals encountered in passive sonar are approximately cyclo-stationary in that a given waveform is repeated at almost a constant rate. The underlying physical process causing the sound is often a complex train of nonidentical impulses. The ability to track the frequency of these pulses over time can be used in such things as detection, classification and target motion analysis. Methods based on high resolution Fourier analysis generally assume that the signal can be represented by a coherent sum over a large number of harmonic components, any one of which can be tracked over time. However there can be advantages in tracking at a time resolution sufficient to detect the pulses in time domain. One is that many harmonics can be tracked in parallel rather than individually. Another is better system identification by detecting the different underlying pulse trains. The contribution here is to evaluate the implications of cyclic summations for the signal-to-noise ratio, and demonstrate its application to frequency tracking and system identification.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129849499","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":"Robust Multiuser Cdma Receivers Using Sequential Decoding","authors":"R. Jana, Lei Wei","doi":"10.1109/ISSPA.1996.615706","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615706","url":null,"abstract":"This paper considers the application of sequential decoding to the detection of data transmitted over the additive white Gaussian noise channel by K synchronous transmitters using direct sequence spread spectrum multiple access (DS-SSMA). A modification of Fano’s sequential decoding metric is derived using Gaussian approximation method. The performance of such a decoder that uses the improved metric for a multiuser system is compared using computer simulations. It is found that the decoder achieves results comparable to the optimal receiver with much reduced complexity.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128462899","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":"Three-Point Search Algorithm for Video Compression","authors":"Z. Cai, V. Tran, A. Bradley","doi":"10.1109/ISSPA.1996.615696","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615696","url":null,"abstract":"Based on the analysis of motion directions, a new search algorithm called Three-Point Search (TPS) is proposed. The main advantages of this method are as follows: 1) searching accuracy of TPS is very close to that of the full search (FS) algorithm that results in optimal matching results; 2) computational complexity is far less than that of FS. For a maximum motion displacement of w pelslframe, this approach needs 5+3w computations to locate the best match. The average computations is far less than 5+3w because a halfway-stop search technique is used. The simulations showed that the average difference in peak signal-to-noise ratio between FS and TPS is around 0.18 dB to the known CIF video sequences.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"52 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124606977","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":"Performance Analysis of Adaptive Algorithms for Envelope-Constrained Filtering","authors":"W. Zheng","doi":"10.1109/ISSPA.1996.615740","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615740","url":null,"abstract":"The performance of the envelope-constrained filter designed by using the newly developed adaptive algorithms will deteriorate when the input signal is contaminated by noise. Two average window techniques are described and then applied to reduce the effect of noise. It is demonstrated that due to use of an averaging scheme, the convergence of the adaptive algorithms can be greatly improved, the misadjustment caused by the noise can be reduced considerably, and the envelope constraints are more easily satisfied by the filter weight estimate, Simulation results are given.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"93 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124664602","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":"Constant Modulus Blihtd Source Separation Technique: a New Approach","authors":"A. Belouchrani, K. Abed-Meraim","doi":"10.1109/ISSPA.1996.615720","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615720","url":null,"abstract":"In this paper, we present a new approach for the blind source separation problem. Recently, several new techniques have emerged for the Multi-Input-Multi-Output (MIMO) blind identification problem [l, 2, 31 which is an important issue in communications. These techniques estimate the transfer function up to a constant matrix. The purpose of the source separation [4,5] consists of the estimation of such matrix and provides an estimate of the emitted signals. The proposed method estimates the mixing matrix by an Input-Output (IO) identification using as inputs a nonlinear version of the estimated sources. Herein, the nonlinear distortion consists of constraining the Modulus of the inputs of the IO-Identification device to a Constant. Hence the name of the proposed technique. This approach presents many benefits, i) Simple implementations (see below) ii) No ill convergence as indicated by simulations, but no proof is available yet. iii) Implementations are possible either in a block processing or in an adaptive fashion. The effectiveness of the proposed method is illustrated by some numerical simulations.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129476346","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 Robust Kalman Filter for Estimatioh and Tracking of a Class of Periodic Discrete Event Processes","authors":"S. D. Elton, B. J. Slocumb","doi":"10.1109/ISSPA.1996.615708","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615708","url":null,"abstract":"This paper discusses a Kalman filter approach to parameter estimation and tracking for a class of discrete event processes. The proposed estimation techniques operate on the recorded event arrival time sequence of a pulse train signal with pulse occurrence times corrupted by timing noise. In adopting a state space approach to signal modelling, a number of real-world conditions are considered and this leads to the formulation of a ICalman filter estimator that is robust to missing and false data, and to signal model mismatch.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129632636","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":"FPGA-Based DSP - It's About Time","authors":"G. Goslin","doi":"10.1109/ISSPA.1996.615736","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615736","url":null,"abstract":"During your last DSP design did you open your box of tricks to find you have too much information to process in an efficient performancekost profile? DSP Systems Architects are becoming more frequently hindered by the lack of off-the-shelf performance obtainable from programmable DSP devices. Advances in multimedia and communications applications have evolved so rapidly that the performance required for these applications exceeds the processing capability that can be obtained from today's most advanced programmable DSP devices. This can lead to the assumption that using multiple DSP processors or a full-custom ASIC are the only possible solution.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114794231","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 New Fast Pitch Search Algorithm Using Line Spectrum Frequency in the Celp Vocoder","authors":"Myungjin Bae, Sangmok Shon, H. Yoo","doi":"10.1109/ISSPA.1996.615676","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615676","url":null,"abstract":"Code Excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 8 kbps. The major drawback of CELP type coders is a large amount of computation. In this paper, we propose a new pitch searching method that preserves the quality of the CELP vocoder reducing computational complexity. The basic idea is that grasps preliminary pitches using the first formant of speech signal and performs pitch search only about the preliminary pitches. As applying the proposed method to the CELP vocoder, we can reduce complexity by 64% in the pitch search.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125997780","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":"Testing Gaussianity of Multivariate Data Using Entropy","authors":"D. R. Iskander, A. Zoubir","doi":"10.1109/ISSPA.1996.615687","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615687","url":null,"abstract":"To date, many methods have been proposed for testing Gaussianity of a random process in both, the time and frequency domains. In this paper we analyse Gaussianity tests based on entropy for univariate and propose a new entropy based test for multivariate data. Although entropy-based tests for Gaussianity have not received much attention in the past, simulation results indicate that the tests are very competitive compared to other methods. Additionally, the entropy based test for Gaussianity of multivariate data has low computational cost which makes it attractive for practical applications.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130815225","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":"Pitch Detection Based On Prototype Waveforms","authors":"I. Burnett, P. M. Gambino","doi":"10.1109/ISSPA.1996.615678","DOIUrl":"https://doi.org/10.1109/ISSPA.1996.615678","url":null,"abstract":"This paper describes a pitch detection algorithm which builds on the ‘Composite Autocorrelation’ method [ 11. By explicitly considering ‘prototype‘ waveforms within the constituent autocorrelation computations, and detecting the pitch at frequent (5ms) intervals, the algorithm produces a robust and continuous pitch track. This leads to a reduced-delay algorithm that overcomes many of the difficulties encountered when using autocorrelation-based techniques in Prototype Waveform (PW) I Waveform Interpolation (WI) coders [2], [3]. In particular, the technique is able to rapidly track the onsets of voiced and unvoiced speech sections. The algorithm is also able to maintain track for both high and low pitch speakers with promising results at this reduced delay.","PeriodicalId":359344,"journal":{"name":"Fourth International Symposium on Signal Processing and Its Applications","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131091774","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}