{"title":"ISAR superresolution imaging based on APES method","authors":"Fulin Su, Yun Jiao","doi":"10.1109/ISSPA.2005.1580992","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580992","url":null,"abstract":"Modern spectral estimation techniques have been applied to ISAR imaging in order to enhance the resolution for many years. The application of APES (Amplitude and Phase EStimation) filtering approach in ISAR imaging is presented. APES is an adaptive filtering approach which is referred to as the amplitude and phase estimation of a sinusoid, for complex spectral estimation. The APES method can yield more accurate spectral estimates with much lower sidelobes and narrower spectral peaks than FFT method and ESPRIT method. Therefore, the higher resolution can be obtained by using APES method instead of FFT method and ESPRIT method when the target flies smoothly. The simulation results show the effectiveness of APES method.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"36 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133085731","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":"Macroblock layer bit-rates control using the histogram based R-D estimation for MPEG-4","authors":"Dong-Wan Seo, Yoonsik Choe","doi":"10.1109/ISSPA.2005.1580243","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580243","url":null,"abstract":"In this paper, we propose the bit-rate control scheme for the block based video coding like H.263(+) or MPEG-4. The proposed scheme is designed to identify the quantization parameter set through the Lagrangian optimization technique using the histogram based R-D estimation. We calculate the quantization parameter set using the Viterbi algorithm to solve the Lagrangian optimization problem considering the Dquant method of H.263(+) or MPEG-4. The proposed scheme improves the video quality by up to 1.5 dB compared with the TMN8 rate control scheme, and it is more effective in the video sources with dynamic activities of bits than the consistent quality approaches.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132133786","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":"Analysis of an MFCC-based audio indexing system for efficient coding of multimedia sources","authors":"O. Mubarak, E. Ambikairajah, J. Epps","doi":"10.1109/ISSPA.2005.1581014","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1581014","url":null,"abstract":"Discrimination between speech and music signals is an important problem in efficient digital radio broadcasting, particularly for variable bit rate applications such as Internet radio. This paper presents a speech/music discrimination system based on a Mel frequency cepstral coefficient (MFCC) front end and a GMM classifier. This system can be used to select the optimum coding scheme for the current frame of an input signal without knowing a priori whether it contains speech-like or music-like characteristics. An analysis of speech and music error rates for different numbers of MFCCs (from 8 to 28) is presented. For the 46 minute evaluation database used in this experiment, an accuracy of up to 97.14% for music and 93.87% for speech can be attained.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130306701","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}
Joao Dovicchi, João Bosco da Mota Alves, Luiz Fernando, Jacintho Maia, E. D. Mattos
{"title":"N-parametric dilation coefficients: a contribution to the compactly supported wavelets construction","authors":"Joao Dovicchi, João Bosco da Mota Alves, Luiz Fernando, Jacintho Maia, E. D. Mattos","doi":"10.1109/ISSPA.2005.1580184","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580184","url":null,"abstract":"In this paper, authors present a formal description of the algorithm for the construction of N-parametric equations, which provides the attainment of any set of dilation coefficients from a set of N angles that sum /2. These coefficients can be used to construct any dimensional Wavelet filter bank to be applied to analysis and reconstruction of specific signal data.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133955862","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":"An improvement to the multiple window spectrogram using quadratic time-frequency distributions","authors":"N. Stevenson, M. Mesbah, B. Boashash","doi":"10.1109/ISSPA.2005.1581022","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1581022","url":null,"abstract":"This paper suggests an improvement to the multiple window (MW) spectrogram. The improvement was observed when redefining the MW spectrogram in a quadratic time– frequency distribution framework. This alternate view permits the MW method to be interpreted as an application of a single ambiguity domain filter, that consists of the average of each window sequence, to the ambiguity representation of the signal. In addition, the MW spectrogram can be improved by extending the ambiguity domain filter along the Doppler lag () axis. This enhanced MW spectrogram has lower variance and bias when estimating several signals in the presence of additive white Gaussian noise, while retaining the desirable properties of the spectrogram such as realness, non–negativity and minimal cross–terms.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129088398","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":"Blind separation of convolutive mixtures of speech signals using linear combination model","authors":"M. Ohata, T. Mukai, K. Matsuoka","doi":"10.1109/ISSPA.2005.1580189","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580189","url":null,"abstract":"In this paper, we propose a blind separation algorithm for convolutive mixture of source signals on the basis of the information-theoretical approach. This approach requires distribution models of the sources. It is difficult to select the models without prior knowledge of sources. In order to resolve the difficulty, we introduce a distribution model with parameters. We construct the parametric model by linearly combining two density functions corresponding to sub- and super-Gaussian distributions. Our algorithm adaptively estimates the parameters and designs a separat- ing filter. We applied the algorithm to convolutive mix- tures of two speeches in a real environment. The result of our experiments shows that our algorithm can improve separation performance.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129141177","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":"Low complexity channel estimation with pilot symbol assisted modulation","authors":"M. Benjillali, L. Szczecinski","doi":"10.1109/ISSPA.2005.1580977","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580977","url":null,"abstract":"Pilot symbol assisted modulation (PSAM) allows the channel estimation in fast-fading channels. The approaches proposed up to now were mainly concerned with the robustness and simplicity of adaptation of the channel estimators. In this paper we analyze the techniques presented in the literature from the complexity and performance points of view. As a result, we show that using linear interpolation the complexity of the channel estimation may be straightforwardly reduced by order of magnitude without compromising the performance. For channels with very fast fading, we propose a new method which offers a simple trade-off between performance and complexity. Numerical simulations are shown to illustrate the analysis.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115139588","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 full search block matching algorithm using frequency domain","authors":"F. Essannouni, R. Thami, A. Salam, D. Aboutajdine","doi":"10.1109/ISSPA.2005.1580999","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580999","url":null,"abstract":"The overwhelming complexity of block matching algorithm using a full search has prompted many companies and academic researchers to propose a myriad of algorithms. The challenge is to decrease the computational complexity of the full search as much as possible without losing too much performance and quality at the output. In this paper, we propose a new and fast algorithm which achieves exactly the same optimal result as the direct full search algorithm. The key idea is to express a robust matching criteria sum square difference (SSD) in terms of cross correlation operations. Speed is obtained from computing the cross correlations in the frequency domain via the Fast Fourier Transform (FFT). We also present a comparative performance analysis, which shows that the proposed method greatly outperforms the state-ofthe-art in frequency-domain motion estimation.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"231 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127571367","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}
F. Cong, Y. Liang, Shaoling Ji, Yifan Hu, Xizhi Shi
{"title":"Blind speech signal separation based on non-stationary and colored characteristics","authors":"F. Cong, Y. Liang, Shaoling Ji, Yifan Hu, Xizhi Shi","doi":"10.1109/ISSPA.2005.1580982","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580982","url":null,"abstract":"Some algorithms based on Second Order Statistics (SOS) succeed in separating the non-stationary or colored mixing signals. Among those algorithms, the nonstationary signals are blocked, or the time delay is considerable for colored signals. The speech signal is non-stationary and colored. Based on the autocorrelation matrix of the delayed mixing signals in each block, a new algorithm to infer mixing speech signals is formulated. Since our algorithm covers both charactes of speech, the convergence of our algorithm needs fewer steps than those algorithms with only one characteristic; what’s more, the speed of our algorithm for separation is even faster than FastICA. Blind Signal Separation (BSS) experiment on speech signals under instantaneous mixing proves the effectiveness of our algorithm.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125390151","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":"Spurious entropy minima for multimodal source separation","authors":"D. Pham, F. Vrins, M. Verleysen","doi":"10.1109/ISSPA.2005.1580190","DOIUrl":"https://doi.org/10.1109/ISSPA.2005.1580190","url":null,"abstract":"This paper presents two approaches for showing that spurious minima of the entropy may exist in the blind source separation context. The first one is based on the calculation of first and second derivative of the output entropy and The second one is based on entropy approximator for multimodal variable having small overlap between the modes. It is shown that spurious entropy minima arise when the source distribution becomes more and more multimodal.","PeriodicalId":385337,"journal":{"name":"Proceedings of the Eighth International Symposium on Signal Processing and Its Applications, 2005.","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-08-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126894046","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}