Pub Date : 2014-07-14DOI: 10.1109/ICASSP.2014.6854771
S. Maresca, P. Braca, J. Horstmann, R. Grasso
In the context of maritime surveillance, low-power HF surface-wave (HFSW) radars have demonstrated to be a cost-effective long-range early-warning sensor for ship detection and tracking. In this work, multi-target tracking and data fusion techniques are applied to live-recorded data from a network of oceanographic HFSW radars installed in the German Bight (North Sea). This experimentation closely follows the one conducted in the Ligurian Sea (Mediterranean Sea) by NATO Science and Technology Organization (STO) Centre for Maritime Research and Experimentation (CMRE) during the Battlespace Preparation 2009 (BP09) campaign. Ship reports from the Automatic Identification System (AIS), recorded from both coastal and satellite-based stations, are exploited as ground truth information and a methodology is applied to classify the fused tracks and to estimate system performances. Preliminary results are presented and discussed, together with an outline for future works.
{"title":"A network of HF surface wave radars for maritime surveillance: Preliminary results in the German Bight","authors":"S. Maresca, P. Braca, J. Horstmann, R. Grasso","doi":"10.1109/ICASSP.2014.6854771","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854771","url":null,"abstract":"In the context of maritime surveillance, low-power HF surface-wave (HFSW) radars have demonstrated to be a cost-effective long-range early-warning sensor for ship detection and tracking. In this work, multi-target tracking and data fusion techniques are applied to live-recorded data from a network of oceanographic HFSW radars installed in the German Bight (North Sea). This experimentation closely follows the one conducted in the Ligurian Sea (Mediterranean Sea) by NATO Science and Technology Organization (STO) Centre for Maritime Research and Experimentation (CMRE) during the Battlespace Preparation 2009 (BP09) campaign. Ship reports from the Automatic Identification System (AIS), recorded from both coastal and satellite-based stations, are exploited as ground truth information and a methodology is applied to classify the fused tracks and to estimate system performances. Preliminary results are presented and discussed, together with an outline for future works.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"369 1","pages":"6077-6081"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"77643651","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854432
V. Alexandratos, M. Bulut, R. Jasinschi
We introduce a mobile system that is able to detect arousal in realtime based on electrocardiogram and electrodermal activity. The system is using an Android smartphone and wearable sensors, which include a smart watch and a heart rate belt that gather skin conductance and heart rate data, respectively. Algorithms for processing the skin conductance and heart rate data, as well as an automated method for labeling the collected `arousal' and `non-arousal' experimental data are developed. Small-scale user tests show 84% 10-fold, 83% between-subject, and 68% new-subject arousal detection accuracy.
{"title":"Mobile real-time arousal detection","authors":"V. Alexandratos, M. Bulut, R. Jasinschi","doi":"10.1109/ICASSP.2014.6854432","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854432","url":null,"abstract":"We introduce a mobile system that is able to detect arousal in realtime based on electrocardiogram and electrodermal activity. The system is using an Android smartphone and wearable sensors, which include a smart watch and a heart rate belt that gather skin conductance and heart rate data, respectively. Algorithms for processing the skin conductance and heart rate data, as well as an automated method for labeling the collected `arousal' and `non-arousal' experimental data are developed. Small-scale user tests show 84% 10-fold, 83% between-subject, and 68% new-subject arousal detection accuracy.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"174 1","pages":"4394-4398"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"77940111","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854330
Tsung-Chuan Ma, Tung-Chien Chen, Liang-Gee Chen
It is impractical to apply a general spike sorting algorithm for every subject because of the individual characteristics of brain signal. Furthermore, extracting more neural activities for higher accuracy of spike sorting requires more input electrodes as well as large power consumption and chip area. Therefore, several practical constraints are considered in this work when implementing a programmable spike sorting hardware with large number of input channels. In this paper, we provide a 128-channel spike detection processor for spike sorting microsystem without compromise of the power efficiency. This chip consumes only 87.02uW and 9.7uW/mm2 of power density, fabricated with 90nm low-leakage CMOS process.
{"title":"Design and implementation of a low power spike detection processor for 128-channel spike sorting microsystem","authors":"Tsung-Chuan Ma, Tung-Chien Chen, Liang-Gee Chen","doi":"10.1109/ICASSP.2014.6854330","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854330","url":null,"abstract":"It is impractical to apply a general spike sorting algorithm for every subject because of the individual characteristics of brain signal. Furthermore, extracting more neural activities for higher accuracy of spike sorting requires more input electrodes as well as large power consumption and chip area. Therefore, several practical constraints are considered in this work when implementing a programmable spike sorting hardware with large number of input channels. In this paper, we provide a 128-channel spike detection processor for spike sorting microsystem without compromise of the power efficiency. This chip consumes only 87.02uW and 9.7uW/mm2 of power density, fabricated with 90nm low-leakage CMOS process.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"151 1","pages":"3889-3892"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"73397891","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854830
Younggwan Kim, Hoirin Kim
Maximum a posterior (MAP) adaptation is one of the popular and powerful methods for obtaining a speaker-specific acoustic model. Basically, MAP adaptation needs a data storage for speaker adaptive (SA) model as much as speaker independent (SI) model needs. Modern speech recognition systems have a huge number of parameters and deal with millions of users. To reduce the data storage for SA models, in this paper, we propose a constrained maximum likelihood estimation-based speaker adaptation with L1 regularization. By the proposed method, we can more efficiently perform the model adjustments for SA models without almost any loss of phone recognition performance than the conventional sparse MAP adaptation method.
{"title":"Constrained MLE-based speaker adaptation with L1 regularization","authors":"Younggwan Kim, Hoirin Kim","doi":"10.1109/ICASSP.2014.6854830","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854830","url":null,"abstract":"Maximum a posterior (MAP) adaptation is one of the popular and powerful methods for obtaining a speaker-specific acoustic model. Basically, MAP adaptation needs a data storage for speaker adaptive (SA) model as much as speaker independent (SI) model needs. Modern speech recognition systems have a huge number of parameters and deal with millions of users. To reduce the data storage for SA models, in this paper, we propose a constrained maximum likelihood estimation-based speaker adaptation with L1 regularization. By the proposed method, we can more efficiently perform the model adjustments for SA models without almost any loss of phone recognition performance than the conventional sparse MAP adaptation method.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"8 1","pages":"6369-6373"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"81996837","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854148
D. Hack, L. Patton, B. Himed
This paper addresses the problem of detecting an unknown rank-one signal using multiple receivers that are uncalibrated in the sense that they each apply an unknown scaling to the received signal, and their respective noise powers are unknown. This problem has been addressed for the case in which the unknown signal can be modeled as a Gaussian random vector. However, that assumption is not applicable to some signal types, such as the constant modulus signals found in radar and communications. For these problems, the signal can be modeled as a deterministic unknown, which is the approach taken here. We derive a generalized likelihood ratio test for this problem under a low signal-to-noise ratio (SNR) assumption. The resulting detector is invariant to relative scalings of the data, and therefore possesses the constant false alarm rate (CFAR) property with respect to the unknown noise powers. Numerical examples show the proposed detector can outperform CFAR detectors derived under the Gaussian assumption.
{"title":"Multichannel detection of an unknown rank-one signal with uncalibrated receivers","authors":"D. Hack, L. Patton, B. Himed","doi":"10.1109/ICASSP.2014.6854148","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854148","url":null,"abstract":"This paper addresses the problem of detecting an unknown rank-one signal using multiple receivers that are uncalibrated in the sense that they each apply an unknown scaling to the received signal, and their respective noise powers are unknown. This problem has been addressed for the case in which the unknown signal can be modeled as a Gaussian random vector. However, that assumption is not applicable to some signal types, such as the constant modulus signals found in radar and communications. For these problems, the signal can be modeled as a deterministic unknown, which is the approach taken here. We derive a generalized likelihood ratio test for this problem under a low signal-to-noise ratio (SNR) assumption. The resulting detector is invariant to relative scalings of the data, and therefore possesses the constant false alarm rate (CFAR) property with respect to the unknown noise powers. Numerical examples show the proposed detector can outperform CFAR detectors derived under the Gaussian assumption.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"50 1","pages":"2987-2991"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"73283355","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854275
Hajar Razaghi, R. Saatchi, D. Burke, A. Offiah
The correlation between Turkey bones' mass/volume ratios and the magnitude and frequency of the largest peak in their vibration frequency spectra was studied. A computer controlled small hammer was used to induce bone vibrations. The study indicated a correlation of 0.87 between the magnitude of the largest peak and mass/volume ratio. The relationship between the frequency of the largest peak in the frequency spectra of the vibration responses and mass/volume ratio was best represented by a second order polynomial while the relationship between the magnitude of the largest peak and mass/volume ratio was best represented by a first order polynomial.
{"title":"An investigation of relationship between bone vibration frequency and its mass-volume ratio","authors":"Hajar Razaghi, R. Saatchi, D. Burke, A. Offiah","doi":"10.1109/ICASSP.2014.6854275","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854275","url":null,"abstract":"The correlation between Turkey bones' mass/volume ratios and the magnitude and frequency of the largest peak in their vibration frequency spectra was studied. A computer controlled small hammer was used to induce bone vibrations. The study indicated a correlation of 0.87 between the magnitude of the largest peak and mass/volume ratio. The relationship between the frequency of the largest peak in the frequency spectra of the vibration responses and mass/volume ratio was best represented by a second order polynomial while the relationship between the magnitude of the largest peak and mass/volume ratio was best represented by a first order polynomial.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"3 1","pages":"3616-3620"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"91053081","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6853773
Dan Wu, Chenyang Yang, Zixiang Xiong
This paper studies the degrees of freedom (DoF) region of the 2 × 2 × 2 interference network, which is comprised of two sources, two relays and two destinations, each with arbitrary number of antennas. We prove that with linear transceivers, the cut-set outer bound can be achieved without any symbol extensions, except for one specific system setup, which has one-DoF gap to the cut-set bound. We show that to achieve the outer-bound, the transceivers include interference avoidance, cancelation, neutralization and alignment, depending on the antenna configuration.
{"title":"Achieving the degrees of freedom of 2×2×2 interference network with arbitrary antenna configurations","authors":"Dan Wu, Chenyang Yang, Zixiang Xiong","doi":"10.1109/ICASSP.2014.6853773","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853773","url":null,"abstract":"This paper studies the degrees of freedom (DoF) region of the 2 × 2 × 2 interference network, which is comprised of two sources, two relays and two destinations, each with arbitrary number of antennas. We prove that with linear transceivers, the cut-set outer bound can be achieved without any symbol extensions, except for one specific system setup, which has one-DoF gap to the cut-set bound. We show that to achieve the outer-bound, the transceivers include interference avoidance, cancelation, neutralization and alignment, depending on the antenna configuration.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"41 1","pages":"1130-1134"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"90563376","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6855107
Ozlem Tugfe Demir, T. E. Tuncer
In this paper, transmit beamformer design for single group multicast scenario is considered. The problem is solved in discrete form where the beamformer phase and amplitude values are selected from finite discrete sets. Original optimization problem is converted to a linear form by introducing new variables. The solution of the equivalent optimization problem is always feasible as long as the total power is above a certain value. The problem in its linear form is guaranteed to return optimum solution. Proposed approach is very effective and the number of bits can be increased to obtain close to optimum continuous phase and amplitude beamformers.
{"title":"Optimum discrete single group multicast beamforming","authors":"Ozlem Tugfe Demir, T. E. Tuncer","doi":"10.1109/ICASSP.2014.6855107","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855107","url":null,"abstract":"In this paper, transmit beamformer design for single group multicast scenario is considered. The problem is solved in discrete form where the beamformer phase and amplitude values are selected from finite discrete sets. Original optimization problem is converted to a linear form by introducing new variables. The solution of the equivalent optimization problem is always feasible as long as the total power is above a certain value. The problem in its linear form is guaranteed to return optimum solution. Proposed approach is very effective and the number of bits can be increased to obtain close to optimum continuous phase and amplitude beamformers.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"87 1","pages":"7744-7748"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85982141","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6855008
Zhiyuan Weng, P. Djurić
In a network, each agent communicates with its neighbors. All the agents have initial observations, and they update their beliefs with the average of the beliefs in their neighborhoods. It is well known that in the long run, the network will reach consensus. However, the agents do not necessarily converge to the global average of the initial observations of all the agents in the network. Instead, the result is always a weighted average. Moreover, it takes infinite time for the process to converge. In this paper, we address regular networks of agents, where each agent (node) has the same number of agents. We propose a method that allows agents in these networks to learn the global average using the history of its local average in finite time.
{"title":"Efficient learning by consensus over regular networks","authors":"Zhiyuan Weng, P. Djurić","doi":"10.1109/ICASSP.2014.6855008","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855008","url":null,"abstract":"In a network, each agent communicates with its neighbors. All the agents have initial observations, and they update their beliefs with the average of the beliefs in their neighborhoods. It is well known that in the long run, the network will reach consensus. However, the agents do not necessarily converge to the global average of the initial observations of all the agents in the network. Instead, the result is always a weighted average. Moreover, it takes infinite time for the process to converge. In this paper, we address regular networks of agents, where each agent (node) has the same number of agents. We propose a method that allows agents in these networks to learn the global average using the history of its local average in finite time.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"79 1","pages":"7253-7257"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"88595243","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 : 2014-07-14DOI: 10.1109/ICASSP.2014.6854383
Zbyněk Koldovský, P. Tichavský
A fast algorithm to solve weighted ℓ1-minimization problems with N × N square “measuring” matrices is proposed. The method is recursive-in-model-order and tracks a homotopy path that goes through solutions of the optimization sub-tasks in the order of 1 through N. It thus yields solutions for all model orders and performs this task faster than the other compared methods. We show applications of this method in sparse linear system identification, in particular, the estimation of sparse target-cancellation filters for audio source separation.
{"title":"A homotopy recursive-in-model-order algorithm for weighted Lasso","authors":"Zbyněk Koldovský, P. Tichavský","doi":"10.1109/ICASSP.2014.6854383","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854383","url":null,"abstract":"A fast algorithm to solve weighted ℓ1-minimization problems with N × N square “measuring” matrices is proposed. The method is recursive-in-model-order and tracks a homotopy path that goes through solutions of the optimization sub-tasks in the order of 1 through N. It thus yields solutions for all model orders and performs this task faster than the other compared methods. We show applications of this method in sparse linear system identification, in particular, the estimation of sparse target-cancellation filters for audio source separation.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"50 1","pages":"4151-4155"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"81770765","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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