{"title":"AM-signal detection in cognitive radios using first-order cyclostationarity","authors":"Yi Zhou, K. Qaraqe, E. Serpedin, O. Dobre","doi":"10.1109/ICASSP.2010.5496087","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496087","url":null,"abstract":"Cognitive radio is regarded as a novel approach for improving the utilization of precious radio spectrum resource. The detection of very low signal-to-noise ratio (SNR) signals with relaxed a priori information on the signal parameters is of high importance to such radios. This paper proposes a detection algorithm based on the first-order cyclostationarity for amplitude modulated (AM) signals that only requires rough information on the signal bandwidth and carrier frequency. A theoretical asymptotic analysis is performed. Simulation results show that the proposed algorithm performs well at low SNRs.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"211 5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122456654","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}
L. Ravichandran, A. Papandreou-Suppappola, A. Spanias, Z. Lacroix, C. Legendre
{"title":"Time-frequency based biological sequence querying","authors":"L. Ravichandran, A. Papandreou-Suppappola, A. Spanias, Z. Lacroix, C. Legendre","doi":"10.1109/ICASSP.2010.5495708","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495708","url":null,"abstract":"We investigate the use of time-frequency (TF) methods to query biological sequences in search of regions of similarity or critical relationships among the sequences. Existing querying approaches are insensitive to repeats, especially in low-complexity regions, and do not provide much support for efficiently querying sub-sequences with inserts and deletes (or gaps). Our approach uses highly-localized basis functions and multiple transformations in the TF plane to map characters in a sequence as well as different properties of a sub-sequence, such as its position in the sequence or number of gaps between sub-sequences. We analyze gapped query-based alignment methods using transformations in the TF plane while demonstrating the method's possible operation in real-time without pre-processing. The algorithm's performance is compared to the widely-accepted BLAST alignment approach, and a significance improvement is observed for queries with repetitive segments.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"50 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122642410","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":"Graph-based regularization for spherical signal interpolation","authors":"Tamara Tosic, Tamara Frossard","doi":"10.1109/ICASSP.2010.5495243","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495243","url":null,"abstract":"This paper addresses the problem of the interpolation of 2-d spherical signals from non-uniformly sampled and noisy data. We propose a graph-based regularization algorithm to improve the signal reconstructed by local interpolation methods such as nearest neighbour or kernel-based interpolation algorithms. We represent the signal as a function on a graph where weights are adapted to the particular geometry of the sphere. We then solve a total variation (TV) minimization problem with a modified version of Chambolle's algorithm. Experimental results with noisy and uncomplete datasets show that the regularization algorithm is able to improve the result of local interpolation schemes in terms of reconstruction quality.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122835009","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":"Spatial and temporal pooling of image quality metrics for perceptual video quality assessment on packet loss streams","authors":"Junyong You, J. Korhonen, A. Perkis","doi":"10.1109/ICASSP.2010.5495313","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495313","url":null,"abstract":"Video streaming through bandwidth-limited channels often suffer from packet losses. Therefore, perceptual quality assessment on video sequences with packet losses is a critical issue in digital video communications. This paper analyzes several image quality metrics and evaluates their applications using spatial and temporal pooling schemes in perceptual video quality assessment for video streams with packet losses. Several approaches using Minkowski summation and averages over different distorted spatial regions and temporal frames to pool the spatial and temporal qualities are evaluated. The experimental results with respect to the subjective video quality measurements demonstrate that the subjects are more sensitive to the most annoying spatial regions and temporal segments when assessing the video quality of the lossy streams.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"18 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122855312","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":"Spoken language translation from parallel speech audio: Simultaneous interpretation as SLT training data","authors":"M. Paulik, A. Waibel","doi":"10.1109/ICASSP.2010.5494998","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5494998","url":null,"abstract":"In recent work, we proposed an alternative to parallel text as translation model (TM) training data: audio recordings of parallel speech (pSp), as it occurs in any communication scenario where interpreters are involved. Although interpretation compares poorly to translation, we reported surprisingly strong translation results for systems based on pSp trained TMs. This work extends the use of pSp as a data source for unsupervised training of all major models involved in statistical spoken language translation. We consider the scenario of speech translation between a resource rich and a resource-deficient language. Our seed models are based on 10h of transcribed audio and parallel text comprised of 100k translated words. With the help of 92h of untranscribed pSp audio, and by taking advantage of the redundancy inherent to pSp (the same information is given twice, in two languages), we report significant improvements for the resource-deficient acoustic, language and translation models.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"32 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122857961","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":"Discriminative training based on an integrated view of MPE and MMI in margin and error space","authors":"E. McDermott, Shinji Watanabe, Atsushi Nakamura","doi":"10.1109/ICASSP.2010.5495106","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495106","url":null,"abstract":"Recent work has demonstrated that the Maximum Mutual Information (MMI) objective function is mathematically equivalent to a simple integral of recognition error, if the latter is expressed as a margin-based Minimum Phone Error (MPE) style error-weighted objective function. This led to the proposal of a general approach to discriminative training based on integrals of MPE-style loss, calculated using “differenced MMI” (dMMI), a finite difference of MMI functionals evaluated at the edges of a margin interval. This article aims to clarify the essence and practical consequences of the new framework. The recently proposed Error-Indexed Forward-Backward Algorithm is used to visualize the close agreement between dMMI and MPE statistics for narrow margin intervals, and to illustrate the flexible control of the weight that can be given to different error levels using broader intervals. New speech recognition results are presented for the MIT OpenCourseWare/MIT-World corpus, showing small performance gains for dMMI compared to MPE for some choices of margin interval. Evaluation with an expanded 44K word trigram language model confirms that dMMI with a narrow margin interval yields the same performance as MPE.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"37 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122933457","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":"Distributed beamforming and mode selection based on instantaneous system throughput","authors":"Jingon Joung, A. H. Sayed","doi":"10.1109/ICASSP.2010.5496120","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496120","url":null,"abstract":"In this paper, we design cooperative beamforming weights for source, relay and destination nodes based on a minimum means-quare-error (MMSE) formulation under network power constraints. We also propose a mode selection procedure based on the instantaneous system throughput. Simulation results indicate that the MMSE cooperative beamforming method with mode selection achieves better performance compared to other beamforming methods: direct beamforming, relay beamforming, and cooperative beamforming without mode selection.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121935626","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. Saeidi, T. Kinnunen, Hamid Reza Mohammadi, R. Rodman, P. Fränti
{"title":"Joint frame and Gaussian selection for text independent speaker verification","authors":"R. Saeidi, T. Kinnunen, Hamid Reza Mohammadi, R. Rodman, P. Fränti","doi":"10.1109/ICASSP.2010.5495576","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495576","url":null,"abstract":"Gaussian selection is a technique applied in the GMM-UBM framework to accelerate score calculation. We have recently introduced a novel Gaussian selection method known as sorted GMM (SGMM). SGMM uses scalar-indexing of the universal background model mean vectors to achieve fast search of the top-scoring Gaussians. In the present work we extend this method by using 2-dimensional indexing, which leads to simultaneous frame and Gaussian selection. Our results on the NIST 2002 speaker recognition evaluation corpus indicate that both the 1- and 2- dimensional SGMMs outperform frame decimation and temporal tracking of top-scoring Gaussians by a wide margin (in terms of Gaussian computations relative to GMM-UBM as baseline).","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116748348","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 gain control for single-carrier communications with uniform quantization at the receiver","authors":"S. Krone, G. Fettweis","doi":"10.1109/ICASSP.2010.5495947","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495947","url":null,"abstract":"The achievable rate of digital communications systems can strongly depend on the analog-to-digital conversion at the receiver. It is hence important to adjust the gain control at the receiver in such a way that the performance degradation due to the analog-to-digital conversion is as small as possible. This paper studies the concept of an optimal gain control to maximize the achievable rate of single-carrier systems that employ analog-to-digital conversion with uniform quantization. Transmission of complex-valued symbols is considered, and a phase offset between transmitter and receiver is taken into account. The optimal gain control derives from the average mutual information between the transmitted symbols and quantized received samples. Allowing for a small tolerance of the achievable rate, it is possible to adequately adjust the gain control even with limited accuracy.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116748612","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":"Wireless source localization based on time of arrival measurement","authors":"E. Xu, Z. Ding, S. Dasgupta","doi":"10.1109/ICASSP.2010.5496191","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496191","url":null,"abstract":"Wireless source localization has found a number of applications in wireless sensor networks. In this work, we investigate source localization based on the practical time of arrival (TOA) measurement model. Unlike most existing works that transform TOA measurement into time differences before processing, we consider the original measurement model and investigate three methods for direct source localization. We also derive the Cramér-Rao lower bound (CRLB) under the TOA model and establish its connection with the CRLB under the more commonly used time-difference of arrival (TDOA) signal model. We present results that illustrate the performance advantage of source localization based on the original TOA model over the commonly used TDOA pre-processing.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"87 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116752349","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}