Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853643
Yao Cheng, Peng Li, M. Haardt
In this contribution, we propose a coordinated transmit beamforming technique for point-to-point multiple-input-multiple-output (MIMO) filter bank based multi-carrier with offset quadrature amplitude modulation (FBMC/OQAM) systems. To enable reliable transmissions when the number of transmit antennas does not exceed the number of receive antennas and the channel is not flat fading, we design a joint and iterative procedure to calculate the precoding matrix and the decoding matrix for each subcarrier. Simulation results show that the proposed algorithm outperforms the existing transmission strategies for MIMO FBMC/OQAM systems. It is also observed that by employing the proposed coordinated beamforming scheme, the MIMO FBMC/OQAM system achieves a similar bit error rate (BER) performance as its orthogonal frequency division multiplexing with the cyclic prefix insertion (CP-OFDM) based counterpart while exhibiting superiority in terms of a higher spectral efficiency, a greater robustness against synchronization errors, and a lower out-of-band radiation.
{"title":"Coordinated beamforming in MIMO FBMC/OQAM systems","authors":"Yao Cheng, Peng Li, M. Haardt","doi":"10.1109/ICASSP.2014.6853643","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853643","url":null,"abstract":"In this contribution, we propose a coordinated transmit beamforming technique for point-to-point multiple-input-multiple-output (MIMO) filter bank based multi-carrier with offset quadrature amplitude modulation (FBMC/OQAM) systems. To enable reliable transmissions when the number of transmit antennas does not exceed the number of receive antennas and the channel is not flat fading, we design a joint and iterative procedure to calculate the precoding matrix and the decoding matrix for each subcarrier. Simulation results show that the proposed algorithm outperforms the existing transmission strategies for MIMO FBMC/OQAM systems. It is also observed that by employing the proposed coordinated beamforming scheme, the MIMO FBMC/OQAM system achieves a similar bit error rate (BER) performance as its orthogonal frequency division multiplexing with the cyclic prefix insertion (CP-OFDM) based counterpart while exhibiting superiority in terms of a higher spectral efficiency, a greater robustness against synchronization errors, and a lower out-of-band radiation.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"74 1","pages":"484-488"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"77780399","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6854306
Ming Li, Xin Li
We propose an ECG based robust human verification system for both healthy and cardiac irregular conditions using the heartbeat level and segment level information fusion. At the heartbeat level, we first propose a novel beat normalization and outlier removal algorithm after peak detection to extract normalized representative beats. Then after principal component analysis (PCA), we apply linear discriminant analysis (LDA) and within-class covariance normalization (WCCN) for beat variability compensation followed by cosine similarity and Snorm as scoring. At the segment level, we adopt the hierarchical Dirichlet process auto-regressive hidden Markov model (HDP-AR-HMM) in the Bayesian non-parametric framework for unsupervised joint segmentation and clustering without any peak detection. It automatically decodes each raw signal into a string vector. We then apply n-gram language model and hypothesis testing for scoring. Combining the aforementioned two subsystems together further improved the performance and outperformed the PCA baseline by 25% relatively on the PTB database.
{"title":"Verification based ECG biometrics with cardiac irregular conditions using heartbeat level and segment level information fusion","authors":"Ming Li, Xin Li","doi":"10.1109/ICASSP.2014.6854306","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854306","url":null,"abstract":"We propose an ECG based robust human verification system for both healthy and cardiac irregular conditions using the heartbeat level and segment level information fusion. At the heartbeat level, we first propose a novel beat normalization and outlier removal algorithm after peak detection to extract normalized representative beats. Then after principal component analysis (PCA), we apply linear discriminant analysis (LDA) and within-class covariance normalization (WCCN) for beat variability compensation followed by cosine similarity and Snorm as scoring. At the segment level, we adopt the hierarchical Dirichlet process auto-regressive hidden Markov model (HDP-AR-HMM) in the Bayesian non-parametric framework for unsupervised joint segmentation and clustering without any peak detection. It automatically decodes each raw signal into a string vector. We then apply n-gram language model and hypothesis testing for scoring. Combining the aforementioned two subsystems together further improved the performance and outperformed the PCA baseline by 25% relatively on the PTB database.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"41 1","pages":"3769-3773"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"80107285","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853664
Iana Iatsun, M. Larabi, C. Fernandez-Maloigne
Saliency is one of the most important features in human visual perception. It is widely used nowadays for perceptually optimizing image processing algorithms. Several models have been proposed for 2D images and only few attempts can be observed for 3D ones. In this paper, we propose a stereoscopic 3D saliency model relying on 2D saliency features jointly with depth obtained from monocular cues. On the one hand, the use of 2D saliency features is justified psychophysically by the similarity observed between 2D and 3D attention maps. On the other hand, 3D perception is significantly based on monocular cues. The validation of our model using state-of-the-art procedures including Kullback-Leibler divergence (KLD), area under the curve (AUC) and correlation coefficient (CC) in comparison with attention maps showed very good performance.
{"title":"Using monocular depth cues for modeling stereoscopic 3D saliency","authors":"Iana Iatsun, M. Larabi, C. Fernandez-Maloigne","doi":"10.1109/ICASSP.2014.6853664","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853664","url":null,"abstract":"Saliency is one of the most important features in human visual perception. It is widely used nowadays for perceptually optimizing image processing algorithms. Several models have been proposed for 2D images and only few attempts can be observed for 3D ones. In this paper, we propose a stereoscopic 3D saliency model relying on 2D saliency features jointly with depth obtained from monocular cues. On the one hand, the use of 2D saliency features is justified psychophysically by the similarity observed between 2D and 3D attention maps. On the other hand, 3D perception is significantly based on monocular cues. The validation of our model using state-of-the-art procedures including Kullback-Leibler divergence (KLD), area under the curve (AUC) and correlation coefficient (CC) in comparison with attention maps showed very good performance.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"28 1","pages":"589-593"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"80429259","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853604
Shinnosuke Takamichi, T. Toda, Graham Neubig, S. Sakti, Satoshi Nakamura
In this paper, we propose a postfilter to compensate modulation spectrum in HMM-based speech synthesis. In order to alleviate over-smoothing effects which is a main cause of quality degradation in HMM-based speech synthesis, it is necessary to consider features that can capture over-smoothing. Global Variance (GV) is one well-known example of such a feature, and the effectiveness of parameter generation algorithm considering GV have been confirmed. However, the quality gap between natural speech and synthetic speech is still large. In this paper, we introduce the Modulation Spectrum (MS) of speech parameter trajectory as a new feature to effectively capture the over-smoothing effect, and we propose a postfilter based on the MS. The MS is represented as a power spectrum of the parameter trajectory. The generated speech parameter sequence is filtered to ensure that its MS has a pattern similar to natural speech. Experimental results show quality improvements when the proposed methods are applied to spectral and F0 components, compared with conventional methods considering GV.
{"title":"A postfilter to modify the modulation spectrum in HMM-based speech synthesis","authors":"Shinnosuke Takamichi, T. Toda, Graham Neubig, S. Sakti, Satoshi Nakamura","doi":"10.1109/ICASSP.2014.6853604","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853604","url":null,"abstract":"In this paper, we propose a postfilter to compensate modulation spectrum in HMM-based speech synthesis. In order to alleviate over-smoothing effects which is a main cause of quality degradation in HMM-based speech synthesis, it is necessary to consider features that can capture over-smoothing. Global Variance (GV) is one well-known example of such a feature, and the effectiveness of parameter generation algorithm considering GV have been confirmed. However, the quality gap between natural speech and synthetic speech is still large. In this paper, we introduce the Modulation Spectrum (MS) of speech parameter trajectory as a new feature to effectively capture the over-smoothing effect, and we propose a postfilter based on the MS. The MS is represented as a power spectrum of the parameter trajectory. The generated speech parameter sequence is filtered to ensure that its MS has a pattern similar to natural speech. Experimental results show quality improvements when the proposed methods are applied to spectral and F0 components, compared with conventional methods considering GV.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"27 1","pages":"290-294"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"76704780","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6855120
M. Cubero-Castan, J. Chanussot, X. Briottet, M. Shimoni, V. Achard
The estimation of surface emissivity and temperature from thermal hyperspectral data is a challenge. Methods that estimate the temperature and emissivity on a pixel composed by one single material exist. However, the estimation of the temperature on a mixed pixel, i.e. a pixel composed by more than one material, is more complex and has scarcely been investigated in the literature. This paper addresses this issue by proposing an estimator which linearizes the Black Body law around the mean temperature of each material. The performance of this estimator is studied using simulated data with different hyperspectral sensor configurations and under various noise conditions. The obtained results are encouraging and show an accuracy on the estimated temperature of 0.5 K while using high spectral resolution sensor.
{"title":"An unmixing-based method for the analysis of thermal hyperspectral images","authors":"M. Cubero-Castan, J. Chanussot, X. Briottet, M. Shimoni, V. Achard","doi":"10.1109/ICASSP.2014.6855120","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855120","url":null,"abstract":"The estimation of surface emissivity and temperature from thermal hyperspectral data is a challenge. Methods that estimate the temperature and emissivity on a pixel composed by one single material exist. However, the estimation of the temperature on a mixed pixel, i.e. a pixel composed by more than one material, is more complex and has scarcely been investigated in the literature. This paper addresses this issue by proposing an estimator which linearizes the Black Body law around the mean temperature of each material. The performance of this estimator is studied using simulated data with different hyperspectral sensor configurations and under various noise conditions. The obtained results are encouraging and show an accuracy on the estimated temperature of 0.5 K while using high spectral resolution sensor.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"20 1","pages":"7809-7813"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"76752129","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6854470
Jose Costa Pereira, J. Luque, Xavier Anguera Miró
This paper addresses the problem of document retrieval based on sentiment polarity criteria. A query based on natural spontaneous speech, expressing an opinion about a certain topic, is used to search a repository of documents containing favorable or unfavorable opinions. The goal is to retrieve documents whose opinions more closely resemble the one in the query. A semantic system based on speech transcripts is augmented with information from full-length text articles. Posterior probabilities extracted from the articles are used to regularize their transcription counterparts. This paper makes three important contributions. First, we introduce a framework for polarity analysis of sentiments that can accommodate combinations of different modalities capable of dealing with the absence of any modality. Second, we show that it is possible to improve average precision on speech transcriptions' sentiment retrieval by means of regularization. Third, we demonstrate the robustness of our approach by training regularizers on one dataset, while performing sentiment retrieval experiments, with substantial gains, on another dataset.
{"title":"Sentiment retrieval on web reviews using spontaneous natural speech","authors":"Jose Costa Pereira, J. Luque, Xavier Anguera Miró","doi":"10.1109/ICASSP.2014.6854470","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854470","url":null,"abstract":"This paper addresses the problem of document retrieval based on sentiment polarity criteria. A query based on natural spontaneous speech, expressing an opinion about a certain topic, is used to search a repository of documents containing favorable or unfavorable opinions. The goal is to retrieve documents whose opinions more closely resemble the one in the query. A semantic system based on speech transcripts is augmented with information from full-length text articles. Posterior probabilities extracted from the articles are used to regularize their transcription counterparts. This paper makes three important contributions. First, we introduce a framework for polarity analysis of sentiments that can accommodate combinations of different modalities capable of dealing with the absence of any modality. Second, we show that it is possible to improve average precision on speech transcriptions' sentiment retrieval by means of regularization. Third, we demonstrate the robustness of our approach by training regularizers on one dataset, while performing sentiment retrieval experiments, with substantial gains, on another dataset.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"59 1","pages":"4583-4587"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"82437553","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853934
M. Abbasi, A. Tadaion, S. Gazor
In this paper we propose an algorithm with reduced complexity for the sphere detection (SD) which is used in multiple input multiple output (MIMO) detection algorithms without any performance degradation. The trade-off between the complexity and the bit error rate is a main challenge in wireless MIMO systems. The maximum likelihood (ML) detector considered as the optimum detector in the literatures. Since the complexity of the naive ML detectors is significantly high, the SD algorithms are proposed to lower the complexity. In this paper, we use the result of the geometrical decoder (GD) proposed in [8] which performs as the ML detector and has lower complexity than SD algorithm. We propose a method to further reduce the complexity of this SD algorithm. We show that the complexity is further reduced by almost 60%, i.e, the number of nodes visited by the proposed SD method is in average 60% less than that of the original one.
{"title":"Reduced complexity sphere decoding using a geometrical approach","authors":"M. Abbasi, A. Tadaion, S. Gazor","doi":"10.1109/ICASSP.2014.6853934","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853934","url":null,"abstract":"In this paper we propose an algorithm with reduced complexity for the sphere detection (SD) which is used in multiple input multiple output (MIMO) detection algorithms without any performance degradation. The trade-off between the complexity and the bit error rate is a main challenge in wireless MIMO systems. The maximum likelihood (ML) detector considered as the optimum detector in the literatures. Since the complexity of the naive ML detectors is significantly high, the SD algorithms are proposed to lower the complexity. In this paper, we use the result of the geometrical decoder (GD) proposed in [8] which performs as the ML detector and has lower complexity than SD algorithm. We propose a method to further reduce the complexity of this SD algorithm. We show that the complexity is further reduced by almost 60%, i.e, the number of nodes visited by the proposed SD method is in average 60% less than that of the original one.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"35 1","pages":"1926-1930"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"81352802","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853713
Sebastian Braun, O. Thiergart, Emanuël Habets
When capturing speech in a multi-talker telecommunication scenario, it is desirable to keep the enhanced signal at an equal loudness level for each speaker. Single-channel automatic gain control systems are not able to adjust the level of different talkers when they are simultaneously active. In this work, an automatic spatial gain control (ASGC) algorithm is proposed that adjusts the directional response of an existing informed spatial filter such that the direct sound of multiple sources can be kept at a constant desired loudness level at the output. The spatial filter additionally reduces diffuse sound and ambient noise. It is shown that the proposed AGSC works well within the tested scenario, and is able to adjust the levels of different speakers even during double talk scenarios.
{"title":"Automatic spatial gain control for an informed spatial filter","authors":"Sebastian Braun, O. Thiergart, Emanuël Habets","doi":"10.1109/ICASSP.2014.6853713","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853713","url":null,"abstract":"When capturing speech in a multi-talker telecommunication scenario, it is desirable to keep the enhanced signal at an equal loudness level for each speaker. Single-channel automatic gain control systems are not able to adjust the level of different talkers when they are simultaneously active. In this work, an automatic spatial gain control (ASGC) algorithm is proposed that adjusts the directional response of an existing informed spatial filter such that the direct sound of multiple sources can be kept at a constant desired loudness level at the output. The spatial filter additionally reduces diffuse sound and ambient noise. It is shown that the proposed AGSC works well within the tested scenario, and is able to adjust the levels of different speakers even during double talk scenarios.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"14 1","pages":"830-834"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"81866035","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853666
Richard Rzeszutek, D. Androutsos
In this paper we investigate methods for propagating automatically generated or user-defined labels through an image using edge-preserving filters. We focus on the domain transform filter as it has been used for propagation purposes in the past. The method we present addresses some of the numerical issues that arise with using the filter directly and also improve on the results by better respecting the underlying image structure during the label propagation. Finally we also demonstrate how a filter-based approach is preferable to using global optimization for interpolating automatically generated sparse features.
{"title":"Label propagation through edge-preserving filters","authors":"Richard Rzeszutek, D. Androutsos","doi":"10.1109/ICASSP.2014.6853666","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853666","url":null,"abstract":"In this paper we investigate methods for propagating automatically generated or user-defined labels through an image using edge-preserving filters. We focus on the domain transform filter as it has been used for propagation purposes in the past. The method we present addresses some of the numerical issues that arise with using the filter directly and also improve on the results by better respecting the underlying image structure during the label propagation. Finally we also demonstrate how a filter-based approach is preferable to using global optimization for interpolating automatically generated sparse features.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"29 1","pages":"599-603"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"82077206","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}
Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6854390
Sunav Choudhary, Naveen Kumar, S. Narayanan, U. Mitra
A strategy for active target detection suitable for the use of mobile agents in a field is presented. In particular, there is an interest in autonomous underwater vehicles. By exploiting notions from group testing, the proposed algorithm decides when to collect new samples depending on whether the mobile agent perceives the sensor measurements correspond to noise or a target pattern. Under suitable assumptions about the field emanated by the target, i.e. the target signature is locally low rank in the field, one can efficiently sample the field to locate the target using O(m log m log n) samples on an n × n grid where m ≪ n is a parameter specifying the group size.
提出了一种适合于现场移动agent使用的主动目标检测策略。特别是，人们对自主水下航行器很感兴趣。通过利用群体测试的概念，该算法根据移动代理是否感知到传感器测量值对应于噪声或目标模式来决定何时收集新样本。在对目标发出的磁场进行适当假设的情况下，即目标信号在磁场中是局部低阶的，人们可以在n × n网格上使用O(m log m log n)个样本对磁场进行有效采样以确定目标的位置，其中m≪n是表示组大小的参数。
{"title":"Active target detection with mobile agents","authors":"Sunav Choudhary, Naveen Kumar, S. Narayanan, U. Mitra","doi":"10.1109/ICASSP.2014.6854390","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854390","url":null,"abstract":"A strategy for active target detection suitable for the use of mobile agents in a field is presented. In particular, there is an interest in autonomous underwater vehicles. By exploiting notions from group testing, the proposed algorithm decides when to collect new samples depending on whether the mobile agent perceives the sensor measurements correspond to noise or a target pattern. Under suitable assumptions about the field emanated by the target, i.e. the target signature is locally low rank in the field, one can efficiently sample the field to locate the target using O(m log m log n) samples on an n × n grid where m ≪ n is a parameter specifying the group size.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"45 1","pages":"4185-4189"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"82230544","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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