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2010 18th European Signal Processing Conference最新文献

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Greedy RLS for sparse filters 稀疏滤波器的贪婪RLS
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.42084
B. Dumitrescu, I. Tabus
{"title":"Greedy RLS for sparse filters","authors":"B. Dumitrescu, I. Tabus","doi":"10.5281/ZENODO.42084","DOIUrl":"https://doi.org/10.5281/ZENODO.42084","url":null,"abstract":"We present an adaptive version of the greedy least squares method for finding a sparse approximate solution, with fixed support size, to an overdetermined linear system. The information updated at each time moment consists of a partial orthogonal triangularization of the system matrix and of partial scalar products of its columns, among them and with the right hand side. Since allowing arbitrary changes of the solution support at each update leads to high computation costs, we have adopted a neighbor permutation strategy that changes at most a position of the support with a new one. Hence, the number of operations is lower than that of the standard RLS. Numerical comparisons with standard RLS in an adaptive FIR identification problem show that the proposed greedy RLS has faster convergence and smaller stationary error.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121578823","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}
引用次数: 5
A blind equalization algorithm for multimodulus square-QAM signals 多模平方qam信号的盲均衡算法
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41887
S. Abrar, A. Nandi
{"title":"A blind equalization algorithm for multimodulus square-QAM signals","authors":"S. Abrar, A. Nandi","doi":"10.5281/ZENODO.41887","DOIUrl":"https://doi.org/10.5281/ZENODO.41887","url":null,"abstract":"We present a generic and an efficient multimodulus family of blind equalization algorithm for use in the higherorder quadrature amplitude modulation based digital communication systems. The proposed algorithm is memoryless Bussgang-type and is based on a stochastic gradient descent of a constrained convex cost function. We show it to be capable of blindly equalizing the channel and recovering the carrier phase simultaneously. We also provide the dynamic convergence analysis of the proposed adaptive algorithm.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125562135","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}
引用次数: 0
Efficient scarf detection prior to face recognition 有效的围巾检测先于人脸识别
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41850
R. Min, Angela D'Angelo, J. Dugelay
{"title":"Efficient scarf detection prior to face recognition","authors":"R. Min, Angela D'Angelo, J. Dugelay","doi":"10.5281/ZENODO.41850","DOIUrl":"https://doi.org/10.5281/ZENODO.41850","url":null,"abstract":"Face occlusion is a very challenging problem in face recognition. The performance of face recognition system can decrease drastically due to the presence of partial occlusion on the face. One approach to overcome this problem is to first pre-classify faces into two classes: the clean face and the occluded face; then faces in different classes are treated by different recognition systems. In this case an algorithm which is able to automatically detect the presence of occlusions on the face will be a useful tool to increase the performances of the system. In this paper we present a scarf detection algorithm. In the experimental results the performances of the algorithm are reported and compared with state of the art systems.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128889320","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}
引用次数: 18
A highly efficient optimization scheme for REMOS-based distant-talking speech recognition 基于远程语音识别的高效优化方案
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41948
R. Maas, A. Sehr, M. Gugat, Walter Kellermann
{"title":"A highly efficient optimization scheme for REMOS-based distant-talking speech recognition","authors":"R. Maas, A. Sehr, M. Gugat, Walter Kellermann","doi":"10.5281/ZENODO.41948","DOIUrl":"https://doi.org/10.5281/ZENODO.41948","url":null,"abstract":"A highly efficient decoding algorithm for the REMOS (REverberation MOdeling for Speech recognition) concept for distant-talking speech recognition as proposed in [1] is suggested to reduce the computational complexity by about two orders of magnitude and thereby allowing for first real-time implementations. REMOS is based on a combined acoustic model consisting of a conventional hidden Markov model (HMM), modeling the clean speech, and a reverberation model. During recognition, the most likely clean-speech and reverberant contributions are estimated by solving an inner optimization problem for logarithmic melspectral (log-melspec) features. In this paper, two approximation techniques for the inner optimization problem are derived. Connected digit recognition experiments confirm that the computational complexity is significantly reduced. Ensuring that the global optima of the inner optimization problem are found, the decoding algorithm based on the proposed approximations even increases the recognition accuracy relative to interior point optimization techniques.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124244244","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}
引用次数: 7
Selective encryption of the MC-EZBC bitstream and residual information MC-EZBC位流和剩余信息的选择性加密
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.42047
H. Hofbauer, A. Uhl
{"title":"Selective encryption of the MC-EZBC bitstream and residual information","authors":"H. Hofbauer, A. Uhl","doi":"10.5281/ZENODO.42047","DOIUrl":"https://doi.org/10.5281/ZENODO.42047","url":null,"abstract":"When selective encryption is used for security in DRM schemes some information of the original bitstream is intentionally left in plain text. This can have various reasons, e.g. generating preview versions for try and buy scenarios. In the case of the MC-EZBC there is also the goal of retaining the scaling capability in the encrypted domain. However, since parts of the bitstream remain in plaintext this information is available to a potential attacker at all times. In this paper we will assess which attacks can be done with this residual information. Consequently we will extend a prior version of selective encryption for the MC-EZBC to include motion vectors.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"238 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124295749","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}
引用次数: 7
Evaluating sensorineural hearing loss with an auditory nerve model using a mean structural similarity measure 使用平均结构相似性测量的听神经模型评估感音神经性听力损失
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41837
Andrew Hines, N. Harte
{"title":"Evaluating sensorineural hearing loss with an auditory nerve model using a mean structural similarity measure","authors":"Andrew Hines, N. Harte","doi":"10.5281/ZENODO.41837","DOIUrl":"https://doi.org/10.5281/ZENODO.41837","url":null,"abstract":"Hearing loss research has traditionally been based on perceptual criteria, speech intelligibility and threshold levels. The development of computational models of the auditory-periphery has allowed experimentation via simulation to provide quantitative, repeatable results at a more granular level than would be practical with clinical research on human subjects. This work seeks to create an objective measure to automate this inspection process and ranks hearing losses based on auditory-nerve discharge patterns. A systematic way of assessing phonemic degradation using the outputs of an auditory nerve model for a range of sensorineural hearing losses would aid in rapid prototyping development of speech-processing algorithms for digital hearing aids. The effect of sensorineural hearing loss (SNHL) on phonemic structure was evaluated in this study using two types of neurograms: temporal fine structure (TFS) and average discharge rate or temporal envelope. The mean structural similarity index (MSSIM) is an objective measure originally developed to assess perceptual image quality. The measure is adapted here for use in measuring the phonemic degradation in neurograms derived from impaired auditory nerve outputs. A full evaluation of the choice of parameters for the metric is presented using a large amount of natural human speech. The metric's boundedness and the results for TFS neurograms indicate it is a superior metric to standard point to point metrics of relative mean absolute error and relative mean squared error.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"18 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114727736","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}
引用次数: 0
A semi-blind base station power estimation algorithm for interference aware HSDPA receivers 基于干扰感知的HSDPA接收机半盲基站功率估计算法
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41949
Clemens Buchacher, J. Wehinger, M. Huemer
{"title":"A semi-blind base station power estimation algorithm for interference aware HSDPA receivers","authors":"Clemens Buchacher, J. Wehinger, M. Huemer","doi":"10.5281/ZENODO.41949","DOIUrl":"https://doi.org/10.5281/ZENODO.41949","url":null,"abstract":"In order to mitigate multipath and inter-cell interference in a mobile terminal, the received signal statistics have to be estimated. In a Code Division Multiple Access (CDMA) system with multiple users, it is difficult to estimate all sources of interference directly. The multipath channel can be estimated using a pilot sequence, but since the pilot is code-multiplexed, it gives no indication for other code powers. We show that assuming a fixed ratio between pilot power and total power has a severe impact on throughput performance if multipath or inter-cell interference are dominant. We develop a low-complexity, semi-blind power estimation algorithm, which is optimized for minimum estimation variance and delay. The algorithm is benchmarked using the interference aware High Speed Downlink Packet Access (HSDPA) type 3i reference implementation as defined by the 3GPP standard.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128456737","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}
引用次数: 0
A comparative evaluation of pitch modification techniques 音高修饰技术的比较评价
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.41855
Thomas Drugman, T. Dutoit
{"title":"A comparative evaluation of pitch modification techniques","authors":"Thomas Drugman, T. Dutoit","doi":"10.5281/ZENODO.41855","DOIUrl":"https://doi.org/10.5281/ZENODO.41855","url":null,"abstract":"This paper addresses the problem of pitch modification, as an important module for an efficient voice transformation system. The Deterministic plus Stochastic Model of the residual signal we proposed in a previous work is compared to TDPSOLA, HNM and STRAIGHT. The four methods are compared through an important subjective test. The influence of the speaker gender and of the pitch modification ratio is analyzed. Despite its higher compression level, the DSM technique is shown to give similar or better results than other methods, especially for male speakers and important ratios of modification. The DSM turns out to be only outperformed by STRAIGHT for female voices.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"63 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127059076","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}
引用次数: 5
Fuzzy contour characterization by subspace based methods of array processing and direct method 基于子空间的阵列处理和直接法模糊轮廓表征
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.42177
Haiping Jiang, J. Marot, C. Fossati, S. Bourennane
{"title":"Fuzzy contour characterization by subspace based methods of array processing and direct method","authors":"Haiping Jiang, J. Marot, C. Fossati, S. Bourennane","doi":"10.5281/ZENODO.42177","DOIUrl":"https://doi.org/10.5281/ZENODO.42177","url":null,"abstract":"Blurred contours are often sought in digital image processing. Most of the existing techniques try to detect blurred contours by fuzzy techniques. In this article, we propose to model the blurred contours by generating the virtual signals from the image, and provide a new viewpoint to detect blurred contour. Especially, the blurred contour with exponential distribution can be presented by three parameters. Then, array processing methods are adopted to estimate these parameters, and optimization method is also considered when the spread parameter is estimated. Experiments finally prove that the proposed methods estimate the parameters of blurred contours accurately and with a reduced computational load.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"47 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127643807","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}
引用次数: 4
BIAS corrections in linear MMSE estimation with large filters 大滤波器线性MMSE估计中的BIAS校正
2010 18th European Signal Processing Conference Pub Date : 2010-08-23 DOI: 10.5281/ZENODO.42214
J. Serra, F. Rubio
{"title":"BIAS corrections in linear MMSE estimation with large filters","authors":"J. Serra, F. Rubio","doi":"10.5281/ZENODO.42214","DOIUrl":"https://doi.org/10.5281/ZENODO.42214","url":null,"abstract":"We investigate optimal bias corrections in the problem of linear minimum mean square error (LMMSE) estimation of a scalar parameter linearly described by a set of Gaussian multidimensional observations. The problem of finding the optimal scaling of a class of LMMSE filter implementations based on the sample covariance matrix (SCM) is addressed. By applying recent results from random matrix theory, the scaling factor minimizing the mean square error (MSE) and depending on both the unknown covariance matrix and its sample estimator is firstly asymptotically analyzed in terms of key scenario parameters, and finally estimated using the SCM. As a main result, a universal scaling factor minimizing the estimator MSE is obtained which dramatically outperforms the conventional LMMSE filter implementation. A Bayesian setting assuming random unknown parameters with known mean and variance is considered in this paper, but exactly the same methodology applies to the classical estimation setup considering deterministic parameters.","PeriodicalId":409817,"journal":{"name":"2010 18th European Signal Processing Conference","volume":"48 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-08-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121231422","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}
引用次数: 6
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