In this paper we focus on the transmission of information signals via chaotic oscillations. To this end, we consider systems which contain generators with additional control loop and could behave chaotically and which dynamics may be controlled using feedback or directional coupling. Below we discuss three schemes of signal transmission and detection using 1) phase or frequency controlled generators, 2) coupled Chua's circuits with the adaptive parameter control, and 3) direction-ally coupled generators extracting binary signal from chaos in a presence of noise. New capabilities of conventional control systems for producing and processing chaotic signals are very promising — both for individual use and for implementation in networks [Shalfeev et al, 94].
{"title":"Signal processing via synchronized chaotic systems with feedback control","authors":"A. Kozlov, V. Shalfeev","doi":"10.5281/zenodo.36116","DOIUrl":"https://doi.org/10.5281/zenodo.36116","url":null,"abstract":"In this paper we focus on the transmission of information signals via chaotic oscillations. To this end, we consider systems which contain generators with additional control loop and could behave chaotically and which dynamics may be controlled using feedback or directional coupling. Below we discuss three schemes of signal transmission and detection using 1) phase or frequency controlled generators, 2) coupled Chua's circuits with the adaptive parameter control, and 3) direction-ally coupled generators extracting binary signal from chaos in a presence of noise. New capabilities of conventional control systems for producing and processing chaotic signals are very promising — both for individual use and for implementation in networks [Shalfeev et al, 94].","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128692412","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}
H. Jozawa, K. Kamikura, K. Yanaka, Hiroshi Watanabe
This paper describes an efficient video coding method using two-stage motion compensation (MC). The proposed MC method employs global MC (CMC) and overlapped block affine MC. CMC is adaptively turned on/off for each macroblock since CMC cannot predict all regions in an image. Simulation results show that the proposed coding method using two-stage MC significantly outperforms H. 263 for sequences with fast motion. Performance improvements in PSNR are about 3–4 dB over H. 263.
{"title":"Video coding using adaptive global MC and local affine MC","authors":"H. Jozawa, K. Kamikura, K. Yanaka, Hiroshi Watanabe","doi":"10.5281/ZENODO.36390","DOIUrl":"https://doi.org/10.5281/ZENODO.36390","url":null,"abstract":"This paper describes an efficient video coding method using two-stage motion compensation (MC). The proposed MC method employs global MC (CMC) and overlapped block affine MC. CMC is adaptively turned on/off for each macroblock since CMC cannot predict all regions in an image. Simulation results show that the proposed coding method using two-stage MC significantly outperforms H. 263 for sequences with fast motion. Performance improvements in PSNR are about 3–4 dB over H. 263.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130634114","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}
It is often assumed that blind separation of dynamically mixed sources can not be accomplished with second order statistics. In this paper it is shown that separation of dynamically mixed sources indeed can be performed using second order statistics only. Two approaches to achieve this separation are presented. The first approach is to use a new criterion, based on second order statistics. The criterion is used in order to derive a gradient based separation algorithm as well modified Newton separation algorithm. The uniqueness of the solution representing separation is also investigated. The other approach is to use System Identification. In this context system identifiability results are presented. Simulations using both the criterion based approach and a Recursive Prediction Error Method are also presented.
{"title":"Source separation using second order statistics","authors":"U. Lindgren, H. Sahlin, H. Broman","doi":"10.5281/ZENODO.36193","DOIUrl":"https://doi.org/10.5281/ZENODO.36193","url":null,"abstract":"It is often assumed that blind separation of dynamically mixed sources can not be accomplished with second order statistics. In this paper it is shown that separation of dynamically mixed sources indeed can be performed using second order statistics only. Two approaches to achieve this separation are presented. The first approach is to use a new criterion, based on second order statistics. The criterion is used in order to derive a gradient based separation algorithm as well modified Newton separation algorithm. The uniqueness of the solution representing separation is also investigated. The other approach is to use System Identification. In this context system identifiability results are presented. Simulations using both the criterion based approach and a Recursive Prediction Error Method are also presented.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"68 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124467927","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}
A novel Fast RLS Algorithm based on the Givens Rotation and developed from an UDUT square-root factorization of autocorrelation matrix is discussed. The algorithm presents excellent numerical properties and requires 14N multiplications and 6N divisions per sampling interval, where N is the linear filter order.
{"title":"A novel givens rotation based fast SQR-RLS algorithm","authors":"A. Carini","doi":"10.5281/ZENODO.36157","DOIUrl":"https://doi.org/10.5281/ZENODO.36157","url":null,"abstract":"A novel Fast RLS Algorithm based on the Givens Rotation and developed from an UDUT square-root factorization of autocorrelation matrix is discussed. The algorithm presents excellent numerical properties and requires 14N multiplications and 6N divisions per sampling interval, where N is the linear filter order.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"214 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133999834","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}
Recently the concept of multitone modulation or OFDM has received much attention. For such a modulation, the dispersiveness of the channel is classically solved by the technique of guard time. In the present paper we investigate the performance of OFDM without guard time but with MIMO equalization. Linear and decision-feedback structures structures are derived for an MMSE criterion and their performance is assessed by means of their steady-state behavior. Symbol rate equalizers following channel matched filters are derived and investigated. It is shown that equalized OFDM outperforms OFDM with guard time.
{"title":"MMSE equalizers for multitone systems without guard time","authors":"L. Vandendorpe","doi":"10.5281/ZENODO.35973","DOIUrl":"https://doi.org/10.5281/ZENODO.35973","url":null,"abstract":"Recently the concept of multitone modulation or OFDM has received much attention. For such a modulation, the dispersiveness of the channel is classically solved by the technique of guard time. In the present paper we investigate the performance of OFDM without guard time but with MIMO equalization. Linear and decision-feedback structures structures are derived for an MMSE criterion and their performance is assessed by means of their steady-state behavior. Symbol rate equalizers following channel matched filters are derived and investigated. It is shown that equalized OFDM outperforms OFDM with guard time.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128116977","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}
Conventional speech scrambling concept is based on permutation of time segments and/or frequency subbands. Although this approach is regarded as an insecure speech encryption method, almost all published scramblers are of that type. W e found out that a linear combination based on Hadamardm atrices instead of conventional permutation gives better cryptographic performances, maintaining all the good features of the scrambling concept. The new scrambling method provides a large key space and a simpler key selection. It attainsnegligible residualintelligibility and higher degree of cryptanalytic immunity. The price of these great improvements is a potential complexity increase. That is why we designed a fast algorithm for the new scrambling method.
{"title":"A new speech scrambling method: Comparative analysis and a fast algorithm","authors":"V. Delić, V. Šenk, V. S. Mibsevic","doi":"10.5281/ZENODO.36005","DOIUrl":"https://doi.org/10.5281/ZENODO.36005","url":null,"abstract":"Conventional speech scrambling concept is based on permutation of time segments and/or frequency subbands. Although this approach is regarded as an insecure speech encryption method, almost all published scramblers are of that type. W e found out that a linear combination based on Hadamardm atrices instead of conventional permutation gives better cryptographic performances, maintaining all the good features of the scrambling concept. The new scrambling method provides a large key space and a simpler key selection. It attainsnegligible residualintelligibility and higher degree of cryptanalytic immunity. The price of these great improvements is a potential complexity increase. That is why we designed a fast algorithm for the new scrambling method.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121263813","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}
In this paper, we describe a hybrid relaxation approach to a graph matching problem, by combining both the discrete and continuous relaxation techniques. Compatibility coefficient, critical factor for both relaxation techniques, is defined in terms of nodes and arcs attributes, and the distance measure between graphs is defined as the inner product of the probability vector and the compatibility vector. The discrete relaxation is used as a preprocessing step to determine the initial matching probabilities, and in the continuous relaxation stage, the final matching probabilities are computed by the gradient projection method, Experimental results show that the proposed algorithm is robust to the corruption of the topologies of the graphs, and the matching probabilities converges very rapidly, alleviating an enormous computational load required for the relaxation process.
{"title":"Graph matching by relaxation technique","authors":"Seong Hak Cheong, Sang Uk Lee","doi":"10.5281/ZENODO.36235","DOIUrl":"https://doi.org/10.5281/ZENODO.36235","url":null,"abstract":"In this paper, we describe a hybrid relaxation approach to a graph matching problem, by combining both the discrete and continuous relaxation techniques. Compatibility coefficient, critical factor for both relaxation techniques, is defined in terms of nodes and arcs attributes, and the distance measure between graphs is defined as the inner product of the probability vector and the compatibility vector. The discrete relaxation is used as a preprocessing step to determine the initial matching probabilities, and in the continuous relaxation stage, the final matching probabilities are computed by the gradient projection method, Experimental results show that the proposed algorithm is robust to the corruption of the topologies of the graphs, and the matching probabilities converges very rapidly, alleviating an enormous computational load required for the relaxation process.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114932083","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}
This paper presents a low rate speech codec (2.4 kb/s) based on a sinusoidal model applied to the excitation signal. A frame classifier in combination with a phase dispersion algorithm allows the same model to be used for voiced as well as unvoiced and transitional sounds. The phase dispersion algorithm significantly improves the perceived quality for all frame classes resulting in more "natural" reconstructed speech. Informal MOS testing indicates that the 2.4 kb/s SEC system achieves MOS scores close to the existing 4 kb/s standards (differences up to 0.2 on the MOS scale) and significantly better than the existing 2.4 kb/s LPC-10 standard (difference of 1.5 on the MOS scale).
{"title":"Subjective performance of spectral excitation coding of speech at 2.4 kb/s","authors":"P. Lupini, V. Cuperman","doi":"10.5281/ZENODO.36343","DOIUrl":"https://doi.org/10.5281/ZENODO.36343","url":null,"abstract":"This paper presents a low rate speech codec (2.4 kb/s) based on a sinusoidal model applied to the excitation signal. A frame classifier in combination with a phase dispersion algorithm allows the same model to be used for voiced as well as unvoiced and transitional sounds. The phase dispersion algorithm significantly improves the perceived quality for all frame classes resulting in more \"natural\" reconstructed speech. Informal MOS testing indicates that the 2.4 kb/s SEC system achieves MOS scores close to the existing 4 kb/s standards (differences up to 0.2 on the MOS scale) and significantly better than the existing 2.4 kb/s LPC-10 standard (difference of 1.5 on the MOS scale).","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125700049","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}
This paper proposes a new method for the efficient computation of the Fisher information matrix of zero-mean complex stationary Gaussian processes. Its complexity (measured by the number of floating point operations) is smaller than the fastest previously available procedure. The key idea exploited is that the Fisher information matrix depends only on the sum of the diagonals of the inverse covariance matrix derivative (with respect to the model parameters), rather than on the whole matrix. To obtain the referred sum, a new efficient technique, built upon the Trench algorithm for computing the inverse of a Toeplitz matrix, is presented.
{"title":"A method for computing the information matrix of stationary Gaussian processes","authors":"J. Bioucas-Dias, J. Leitão","doi":"10.5281/ZENODO.35995","DOIUrl":"https://doi.org/10.5281/ZENODO.35995","url":null,"abstract":"This paper proposes a new method for the efficient computation of the Fisher information matrix of zero-mean complex stationary Gaussian processes. Its complexity (measured by the number of floating point operations) is smaller than the fastest previously available procedure. The key idea exploited is that the Fisher information matrix depends only on the sum of the diagonals of the inverse covariance matrix derivative (with respect to the model parameters), rather than on the whole matrix. To obtain the referred sum, a new efficient technique, built upon the Trench algorithm for computing the inverse of a Toeplitz matrix, is presented.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"17 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124836471","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}
In this paper, we address the problem of estimating the frequency of a sinusoidal signal with random, lowpass amplitude. We propose to use MUSIC and ESPRIT frequency estimators as if the signal had a constant amplitude. The aim of the paper is to analyze the degradation of performance induced by the aforementioned mismodelling. Unified expressions for the bias and variances of the MUSIC and ESPRIT frequency estimators are derived under the hypothesis of small bandwidth of the signal envelope. Numerical simulations illustrate the agreement between theoretical and empirical results and study the influence of the envelope bandwidth onto the frequency estimation performance.
{"title":"Robustness analysis of music and esprit frequency estimators for sinusoidal signals with time-varying amplitude","authors":"O. Besson, P. Stoica","doi":"10.5281/ZENODO.35975","DOIUrl":"https://doi.org/10.5281/ZENODO.35975","url":null,"abstract":"In this paper, we address the problem of estimating the frequency of a sinusoidal signal with random, lowpass amplitude. We propose to use MUSIC and ESPRIT frequency estimators as if the signal had a constant amplitude. The aim of the paper is to analyze the degradation of performance induced by the aforementioned mismodelling. Unified expressions for the bias and variances of the MUSIC and ESPRIT frequency estimators are derived under the hypothesis of small bandwidth of the signal envelope. Numerical simulations illustrate the agreement between theoretical and empirical results and study the influence of the envelope bandwidth onto the frequency estimation performance.","PeriodicalId":282153,"journal":{"name":"1996 8th European Signal Processing Conference (EUSIPCO 1996)","volume":"122 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116036696","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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