{"title":"Video summarization using a visual attention model","authors":"Sophie Marat, M. Guironnet, D. Pellerin","doi":"10.5281/ZENODO.40569","DOIUrl":"https://doi.org/10.5281/ZENODO.40569","url":null,"abstract":"This paper presents a method of video summarization based on a visual attention model. The visual attention model is a bottom-up one composed of two parallel ways. A static way, biologically inspired, which highlights salient objects. A dynamic way which gives information about moving objects. A three steps summary method is then presented. The first step is the choice between the two kinds (static and dynamic) of saliency maps given by the attention model. The second step is the selection of keyframes. An “attention variation curve” which highlights changes on frames content during the video is introduced. Keyframes are selected on this variation attention curve. To evaluate the summary a reference summary is built and a comparison method is proposed. The results provide a quantitative analysis and show the efficiency of the video summarization method.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132663694","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. Antonacci, L. Gerosa, A. Sarti, S. Tubaro, G. Valenzise
{"title":"Sound-based classification of objects using a robust fingerprinting approach","authors":"F. Antonacci, L. Gerosa, A. Sarti, S. Tubaro, G. Valenzise","doi":"10.5281/ZENODO.40679","DOIUrl":"https://doi.org/10.5281/ZENODO.40679","url":null,"abstract":"Tangible Acoustic Interfaces (TAIs) are interaction devices that are able to localize the interaction point on a solid surface. Their advantages over traditional interaction devices (touch screens, touch pads, etc.) is in the fact that actual acoustic (vibrational) signals are acquired by contact sensors. This opens the way to interaction classification and recognition. With this application in mind, this paper approaches the problem of classifying the interaction object from the acquired sounds. We focus on continuous interaction noise, which we classify through a “fingerprinting” approach: features are extracted from the acquired signals and matched against pre-computed features. More sophisticated solutions can be devised for the problem of the classification of noiselike sounds but our approach has the advantage of being computationally simple and can be profitably implemented in real-time.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134643688","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":"Modified CELP coder using root cepstrum","authors":"V. Abolghasemi, H. Marvi","doi":"10.5281/ZENODO.40485","DOIUrl":"https://doi.org/10.5281/ZENODO.40485","url":null,"abstract":"Due to increasing demand for speech communications, efficient techniques in low-rate speech coding are of interest. In this paper a new compression technique using root cepstral analysis has been proposed. Implementing the proposed method causes the coder to deal with root cepstrum coefficients instead of speech samples. The main idea in using root cepstrum analysis is that some of the trivial coefficients can be ignored to send toward the decoder. Although it leaves a little degradation in the quality of decoded speech signal, considerable reduction in the total bit-rate is achieved. Moreover it has the advantage of adjustability which can be used to optimize the coding procedure. The experimental results confirm the ability of the proposed method in speech coding problems.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133676275","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}
D. Kumar, P. Carvalho, M. Antunes, J. Henriques, R. Schmidt, J. Habetha
{"title":"Near real time noise detection during heart sound acquisition","authors":"D. Kumar, P. Carvalho, M. Antunes, J. Henriques, R. Schmidt, J. Habetha","doi":"10.5281/ZENODO.40486","DOIUrl":"https://doi.org/10.5281/ZENODO.40486","url":null,"abstract":"Heart sound is a valuable biosignal for early detection of a large set of cardiac diseases. Ambient and physiological noise interference is one of the most usual and high probable incidents during heart sound acquisition. It tends to change the prominent and crucial characteristics of heart sound which may possess important information for heart disease diagnosis. In this paper, we propose a new method to detect ambient and internal body noises combined with heart sound. The algorithm is based upon the periodic nature of heart sounds and physiologically inspired criteria. A small segment of clean heart sound exhibiting periodicity in time as well as in the frequency domain is first detected and applied as a reference signal in sorting noise from the sound. The achieved average sensitivity and specificity are 94% and 97%, respectively.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129820429","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":"Fast adaptation of frequency-domain volterra filters using inherent recursions of iterated coefficient updates","authors":"M. Zeller, Walter Kellermann","doi":"10.5281/ZENODO.40530","DOIUrl":"https://doi.org/10.5281/ZENODO.40530","url":null,"abstract":"Adaptive Volterra filters are a popular model for compensating distortions caused by nonlinear structures with memory such as low-quality loudspeakers. This paper proposes a fast version of the recently investigated repeated coefficient updates for the partitioned block frequency-domain adaptive Volterra filter. Exploiting inherent recursions of the iteration procedure yields an efficient realization with a very low additional complexity compared to the usual LMS adaptation. Experimental results for both noise and speech demonstrate a significant acceleration of the filter convergence and overall echo cancellation for realistic nonlinear AEC scenarios.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"21 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132586102","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":"Asymptotic generalized eigenvalue distribution of block Toeplitz matrices and application to space-time beamforming","authors":"M. Oudin, J. Delmas","doi":"10.5281/ZENODO.40705","DOIUrl":"https://doi.org/10.5281/ZENODO.40705","url":null,"abstract":"In many detection applications, the main performance criterion is the Signal to Interference plus Noise Ratio (SINR). After linear filtering, the optimal SINR corresponds to the maximum value of a Rayleigh quotient, which can be interpreted as the largest generalized eigenvalue of two covariance matrices. In this paper, we extend the Szegö's theorem for the generalized eigenvalues of Hermitian block Toeplitz matrices derived under the assumption of absolutely summable sequences. Then, we apply this result to wideband spacetime beamforming performance analysis where the optimal SINR can be interpreted as the largest generalized eigenvalue of a block Toeplitz matrices' pair. We show that the optimal space-time SINR converges to an upper bound that can be interpreted as an optimal zero-bandwidth spatial SINR and interpret this result for several jamming scenarios.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"140 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133135257","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 stochastic model for a new robust NLMS algorithm","authors":"L. Vega, H. Rey, J. Benesty, S. Tressens","doi":"10.5281/ZENODO.40281","DOIUrl":"https://doi.org/10.5281/ZENODO.40281","url":null,"abstract":"We present a stochastic model for a new recently proposed robust NLMS algorithm. Under very standard and reasonable assumptions we show that the algorithm converges to the true unknown system in a mean square sense. With the aggregate of more restrictive, but standard, assumptions we can build a model for the transient behavior of the algorithm. The model can take into account the presence of impulsive noise. Finally we also present simulations results which show the excellent agreement with the model.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131819807","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":"Optimal joint source channel coding for scalable video transmission over wireless channels","authors":"N. Ramzan, E. Izquierdo","doi":"10.5281/ZENODO.40342","DOIUrl":"https://doi.org/10.5281/ZENODO.40342","url":null,"abstract":"In this paper, a robust and novel approach for optimal bit allocation between source and channel coding is proposed. The proposed approach consists of a wavelet-based scalable video coding framework and a forward error correction method based on the serial concatenation of LDPC codes and turbo codes. Turbo codes shows good performance at low signal to noise ratios but LDPC outperforms turbo codes at high signal to noise ratios. So the concatenation of LDPC and TC enhances the performance at both low and high signal to noise ratios. The scheme reduces the video distortion at the decoder under band-with constraints. The reduction is achieved by efficiently protecting the different quality layers from channel errors. Furthermore, an efficient decoding algorithm is proposed that reduces the decoding complexity of channel decoder. Experimental results clearly show that the proposed approach outperforms conventional forward error correction techniques.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"234 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134023212","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}
R. Vicen-Bueno, M. Jarabo-Amores, M. Rosa-Zurera, D. Mata-Moya, R. Gil-Pita
{"title":"Robustness with respect to the signal-to-noise ratio of MLP-based detectors in Weibull clutter","authors":"R. Vicen-Bueno, M. Jarabo-Amores, M. Rosa-Zurera, D. Mata-Moya, R. Gil-Pita","doi":"10.5281/ZENODO.40558","DOIUrl":"https://doi.org/10.5281/ZENODO.40558","url":null,"abstract":"The Neyman-Pearson detector can be approximated by MultiLayer Perceptrons (MLPs) trained in a supervised way to minimize the Mean Square Error. The detection of a known target in a Weibull-distributed clutter and white Gaussian noise is considered. Because of the difficulty to obtain analytical expressions for the optimum detector under this environment, a suboptimum detector like the Target Sequence Known A Priori (TSKAP) detector is taken as reference. A study of the MLP size shows as a low complexity MLP-based detector trained with the Levenberg-Marquardt algorithm to minimize the MSE is able to obtain good performances. Low performance improvement is achieved for greater sizes than 20 hidden neurons. The MLP-based detector is better than the TSKAP one, even for very low complexity MLPs (6 inputs, 5 hidden neurons and 1 output). Moreover, it is demonstrated empirically that both detectors are robust with respect to changes in the target parameters (signal to noise ratio). So, MLP-based detectors are proposed to detect known targets in Weibull-distributed clutter plus white Gaussian noise.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134094706","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":"Bandwidth extension of narrowband speech based on blind model adaptation","authors":"Sheng Yao, C. Chan","doi":"10.5281/ZENODO.40685","DOIUrl":"https://doi.org/10.5281/ZENODO.40685","url":null,"abstract":"Traditional telephone transmission network has speech frequency upper-limit below 4 kHz. The narrowband telephone speech (0-4 kHz) sounds muffled as compared with the original wideband speech (0-8 kHz). Artificial bandwidth extension is an economical way of enhancing the quality of narrowband speech without modifying the infrastructure of the network. Existing bandwidth extension methods usually include off-line learning phase and on-line enhancing phase. The performance of these systems depends largely on the consistency of wideband training data and actual narrowband input data. In real situation, input speeches usually mismatch with off-line training speeches, leading to serious model errors. To avoid the data mismatch, we propose a method based on blind adaptation of linear dynamic model. The benefit of our method is the exclusion of off-line training phase and experiment results show that our systems is comparable with those data-oriented systems in the measurements of highband spectral distortion. When data mismatch occurs, our system outperforms those systems.","PeriodicalId":176384,"journal":{"name":"2007 15th European Signal Processing Conference","volume":"264 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-09-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133599567","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}