Pub Date : 2014-07-14DOI: 10.1109/ICASSP.2014.6854095
F. Verde, A. Scaglione, D. Darsena, G. Gelli
In this paper, we propose an opportunistic amplify-and-forward relaying scheme for a cognitive radio network, which is aimed at allowing a secondary user (SU) to transmit over the same time-frequency slot of a primary user (PU). In our scheme, the SU amplifies and transmits the PU signal it receives, by using as relaying gain the information symbols that the SU wishes to transmit towards its own secondary receiver. The information theoretic limits of the proposed protocol are investigated by showing that, in some operative conditions of practical interest, the SU can embed its information symbols in the PU signal, without violating the cognitive radio principle of protecting the PU transmission and, at the same time, by attaining low transmission rates.
{"title":"An amplify-and-forward scheme for cognitive radios","authors":"F. Verde, A. Scaglione, D. Darsena, G. Gelli","doi":"10.1109/ICASSP.2014.6854095","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854095","url":null,"abstract":"In this paper, we propose an opportunistic amplify-and-forward relaying scheme for a cognitive radio network, which is aimed at allowing a secondary user (SU) to transmit over the same time-frequency slot of a primary user (PU). In our scheme, the SU amplifies and transmits the PU signal it receives, by using as relaying gain the information symbols that the SU wishes to transmit towards its own secondary receiver. The information theoretic limits of the proposed protocol are investigated by showing that, in some operative conditions of practical interest, the SU can embed its information symbols in the PU signal, without violating the cognitive radio principle of protecting the PU transmission and, at the same time, by attaining low transmission rates.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"36 1","pages":"2724-2728"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"83833025","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.2014.6854501
K. Yatabe, Yasuhiro Oikawa
An effective way to understand the behavior of a sound field is to visualize it. An optical measurement method is a suitable option for this as it enables contactless non-destructive measurement. After measuring a sound field, interpolation of the data is necessary for a smooth visualization. However, conventional interpolation methods cannot provide a physically meaningful result especially when the condition of the measurement causes moiré effect. In this paper, a special interpolation method for an optically visualized sound field based on the Kirchhoff-Helmholtz integral equation is proposed.
{"title":"PDE-based interpolation method for optically visualized sound field","authors":"K. Yatabe, Yasuhiro Oikawa","doi":"10.1109/ICASSP.2014.6854501","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854501","url":null,"abstract":"An effective way to understand the behavior of a sound field is to visualize it. An optical measurement method is a suitable option for this as it enables contactless non-destructive measurement. After measuring a sound field, interpolation of the data is necessary for a smooth visualization. However, conventional interpolation methods cannot provide a physically meaningful result especially when the condition of the measurement causes moiré effect. In this paper, a special interpolation method for an optically visualized sound field based on the Kirchhoff-Helmholtz integral equation is proposed.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"37 1","pages":"4738-4742"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"89317943","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.2014.6854098
Muhammad R. A. Khandaker, Y. Rong
In this paper, we consider a dual-hop multicasting multiple-input multiple-output (MIMO) relay system where multiple transmitters multicast their own messages to a group of receivers with the aid of a relay node, and all nodes are equipped with multiple antennas. We aim at minimizing the maximal MSE of the signal waveform estimation among all receivers subjecting to power constraints at the transmitters and the relay node. We propose a low complexity solution for the problem under some mild approximation. In particular, we show that under (moderately) high signal-to-noise ratio (SNR) assumption, the min-max optimization problem can be solved using the semidefinite programming (SDP) technique. Numerical simulations demonstrate the effectiveness of the proposed algorithm.
{"title":"Simplified MIMO relay design for multicasting from multiple-sources","authors":"Muhammad R. A. Khandaker, Y. Rong","doi":"10.1109/ICASSP.2014.6854098","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854098","url":null,"abstract":"In this paper, we consider a dual-hop multicasting multiple-input multiple-output (MIMO) relay system where multiple transmitters multicast their own messages to a group of receivers with the aid of a relay node, and all nodes are equipped with multiple antennas. We aim at minimizing the maximal MSE of the signal waveform estimation among all receivers subjecting to power constraints at the transmitters and the relay node. We propose a low complexity solution for the problem under some mild approximation. In particular, we show that under (moderately) high signal-to-noise ratio (SNR) assumption, the min-max optimization problem can be solved using the semidefinite programming (SDP) technique. Numerical simulations demonstrate the effectiveness of the proposed algorithm.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"21 1","pages":"2739-2743"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"83222251","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.2014.6854444
Bing Bu, C. Bao, Mao-shen Jia, Rong Zhu
This paper describes a design approach of Ambisonic reproduction system based on dynamic gain parameters (DGP). In the conventional approaches, the fixed gain parameters are often optimized to minimize the overall objective function for whole 360° sound stage. The proposed approach has an advantage that the gain parameters vary with angles of source objects. The problem of optimization tradeoff among different angles is overcome by DGP, which achieves an optimal solution in each position. Source localizations of the B-Format signals were estimated in frequency bands in order to match the corresponding gain parameters. For the synthesized signals, the process was simplified by the given spatial information. Using the head-related transfer function (HRTF) analysis, the proposed approach was found to be significantly better than reference approaches in interaural time difference (ITD) and interaural level difference (ILD).
{"title":"The design of Ambisonic reproduction system based on dynamic gain parameters","authors":"Bing Bu, C. Bao, Mao-shen Jia, Rong Zhu","doi":"10.1109/ICASSP.2014.6854444","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854444","url":null,"abstract":"This paper describes a design approach of Ambisonic reproduction system based on dynamic gain parameters (DGP). In the conventional approaches, the fixed gain parameters are often optimized to minimize the overall objective function for whole 360° sound stage. The proposed approach has an advantage that the gain parameters vary with angles of source objects. The problem of optimization tradeoff among different angles is overcome by DGP, which achieves an optimal solution in each position. Source localizations of the B-Format signals were estimated in frequency bands in order to match the corresponding gain parameters. For the synthesized signals, the process was simplified by the given spatial information. Using the head-related transfer function (HRTF) analysis, the proposed approach was found to be significantly better than reference approaches in interaural time difference (ITD) and interaural level difference (ILD).","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"61 1","pages":"4453-4457"},"PeriodicalIF":0.0,"publicationDate":"2014-07-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"90338191","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 : 2014-06-02DOI: 10.1109/ICASSP.2014.6854636
R. Carrillo, J. McEwen, Y. Wiaux
In recent works, compressed sensing and convex optimization techniques have been applied to radio-interferometric imaging showing the potential to outperform state-of-the-art imaging algorithms in the field. We review our latest contributions, which leverage the versatility of convex optimization to both handle realistic continuous visibilities and offer a highly parallelizable structure paving the way to significant acceleration of the reconstruction and high-dimensional data scalability. The new algorithmic structure, promoted in a new software PURIFY (beta version), relies on the simultaneous-direction method of multipliers (SDMM). The performance of various sparsity priors is evaluated through simulations in the continuous visibility setting, confirming the superiority of our recent average sparsity approach SARA.
{"title":"Purify: A new algorithmic framework for next-generation radio-interferometric imaging","authors":"R. Carrillo, J. McEwen, Y. Wiaux","doi":"10.1109/ICASSP.2014.6854636","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854636","url":null,"abstract":"In recent works, compressed sensing and convex optimization techniques have been applied to radio-interferometric imaging showing the potential to outperform state-of-the-art imaging algorithms in the field. We review our latest contributions, which leverage the versatility of convex optimization to both handle realistic continuous visibilities and offer a highly parallelizable structure paving the way to significant acceleration of the reconstruction and high-dimensional data scalability. The new algorithmic structure, promoted in a new software PURIFY (beta version), relies on the simultaneous-direction method of multipliers (SDMM). The performance of various sparsity priors is evaluated through simulations in the continuous visibility setting, confirming the superiority of our recent average sparsity approach SARA.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"64 1","pages":"5407-5411"},"PeriodicalIF":0.0,"publicationDate":"2014-06-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"88927692","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.2014.6854272
A. Lanatà, A. Greco, G. Valenza, E. Scilingo
This paper reports on results of a pattern recognition technique for classifying pathological mental states of bipolar disorders using information gathered from the electrodermal response. The rationale behind this work is that the autonomic nervous system dynamics, non-invasively quantified through the electrodermal response processing, is altered by the specific mood state. Starting from the hypothesis that bipolar disorders are associated with affective dysfunctions, we processed data gathered from four bipolar patients through eleven experimental trials while an ad-hoc emotional stimulation is administered. Intra- and inter-subject variability were investigated. We show that, using a deconvolution-based approach to estimate sympathetic ANS markers and simple k-Nearest Neighbor algorithms, the proposed methodology is able to discern up to three mood states such as depression, hypo-mania, and euthymia with an average intra-subject accuracy greater than 98% and inter-subject accuracy greater than 82%.
{"title":"A pattern recognition approach based on electrodermal response for pathological mood identification in bipolar disorders","authors":"A. Lanatà, A. Greco, G. Valenza, E. Scilingo","doi":"10.1109/ICASSP.2014.6854272","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854272","url":null,"abstract":"This paper reports on results of a pattern recognition technique for classifying pathological mental states of bipolar disorders using information gathered from the electrodermal response. The rationale behind this work is that the autonomic nervous system dynamics, non-invasively quantified through the electrodermal response processing, is altered by the specific mood state. Starting from the hypothesis that bipolar disorders are associated with affective dysfunctions, we processed data gathered from four bipolar patients through eleven experimental trials while an ad-hoc emotional stimulation is administered. Intra- and inter-subject variability were investigated. We show that, using a deconvolution-based approach to estimate sympathetic ANS markers and simple k-Nearest Neighbor algorithms, the proposed methodology is able to discern up to three mood states such as depression, hypo-mania, and euthymia with an average intra-subject accuracy greater than 98% and inter-subject accuracy greater than 82%.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"24 1","pages":"3601-3605"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"73565082","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.2014.6853812
I. Edjekouane, C. Plapous, C. Quinquis, S. Meunier
In this paper, we propose a new approach for the subjective assessment of the loudness of complex audio signals such as speech or music. This two-stage approach makes it possible to study the influence on loudness of the frequency bandwidth and of different kinds of codecs. In the first stage, the individual loudness function of each subject is estimated using a specific 100-point response scale. In the second stage, the subject evaluates the loudness of each processed sample, by filtering or coding/decoding, using the same scale. The loudness obtained in terms of points is then converted in loudness levels in terms of phons using the estimated individual loudness function. Results show that loudness increases with the bandwidth extension up to super-wideband. Similar behavior is observed when codecs are applied.
{"title":"Speech and audio loudness depending on telephone audio bandwidth and codec — A subjective testing approach","authors":"I. Edjekouane, C. Plapous, C. Quinquis, S. Meunier","doi":"10.1109/ICASSP.2014.6853812","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853812","url":null,"abstract":"In this paper, we propose a new approach for the subjective assessment of the loudness of complex audio signals such as speech or music. This two-stage approach makes it possible to study the influence on loudness of the frequency bandwidth and of different kinds of codecs. In the first stage, the individual loudness function of each subject is estimated using a specific 100-point response scale. In the second stage, the subject evaluates the loudness of each processed sample, by filtering or coding/decoding, using the same scale. The loudness obtained in terms of points is then converted in loudness levels in terms of phons using the estimated individual loudness function. Results show that loudness increases with the bandwidth extension up to super-wideband. Similar behavior is observed when codecs are applied.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"18 1","pages":"1325-1329"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"73662599","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.2014.6855057
Z. Duan, Bryan Pardo, L. Daudet
We propose a novel cepstral representation called the uniform discrete cepstrum (UDC) to represent the timbre of sound sources in a sound mixture. Different from ordinary cepstrum and MFCC which have to be calculated from the full magnitude spectrum of a source after source separation, UDC can be calculated directly from isolated spectral points that are likely to belong to the source in the mixture spectrum (e.g., non-overlapping harmonics of a harmonic source). Existing cepstral representations that have this property are discrete cepstrum and regularized discrete cepstrum, however, compared to the proposed UDC, they are not as effective and are more complex to compute. The key advantage of UDC is that it uses a more natural and locally adaptive regularizer to prevent it from overfitting the isolated spectral points. We derive the mathematical relations between these cepstral representations, and compare their timbre modeling performances in the task of instrument recognition in polyphonic audio mixtures. We show that UDC and its mel-scale variant MUDC significantly outperform all the other representations.
{"title":"A novel cepstral representation for timbre modeling of sound sources in polyphonic mixtures","authors":"Z. Duan, Bryan Pardo, L. Daudet","doi":"10.1109/ICASSP.2014.6855057","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855057","url":null,"abstract":"We propose a novel cepstral representation called the uniform discrete cepstrum (UDC) to represent the timbre of sound sources in a sound mixture. Different from ordinary cepstrum and MFCC which have to be calculated from the full magnitude spectrum of a source after source separation, UDC can be calculated directly from isolated spectral points that are likely to belong to the source in the mixture spectrum (e.g., non-overlapping harmonics of a harmonic source). Existing cepstral representations that have this property are discrete cepstrum and regularized discrete cepstrum, however, compared to the proposed UDC, they are not as effective and are more complex to compute. The key advantage of UDC is that it uses a more natural and locally adaptive regularizer to prevent it from overfitting the isolated spectral points. We derive the mathematical relations between these cepstral representations, and compare their timbre modeling performances in the task of instrument recognition in polyphonic audio mixtures. We show that UDC and its mel-scale variant MUDC significantly outperform all the other representations.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"72 1","pages":"7495-7499"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"74061692","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.2014.6854985
Julien van Hout, L. Ferrer, D. Vergyri, N. Scheffer, Yun Lei, V. Mitra, S. Wegmann
State-of-the-art calibration and fusion approaches for spoken term detection (STD) systems currently rely on a multi-pass approach where the scores are calibrated, then fused, and finally re-calibrated to obtain a single decision threshold across keywords. While the above techniques are theoretically correct, they rely on meta-parameter tuning and are prone to over-fitting. This study presents an efficient and effective score calibration technique for keyword detection that is based on the logistic regression calibration approach commonly used in forensic speaker identification. The technique applies seamlessly to both single systems and to system fusion, and enables optimization for specific keyword detection evaluation functions. We run experiments on a Vietnamese STD task, comparing the technique with more empirical calibration and fusion schemes and demonstrate that we can achieve comparable or better performance in terms of the NIST ATWV metric with a more elegant solution.
{"title":"Calibration and multiple system fusion for spoken term detection using linear logistic regression","authors":"Julien van Hout, L. Ferrer, D. Vergyri, N. Scheffer, Yun Lei, V. Mitra, S. Wegmann","doi":"10.1109/ICASSP.2014.6854985","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854985","url":null,"abstract":"State-of-the-art calibration and fusion approaches for spoken term detection (STD) systems currently rely on a multi-pass approach where the scores are calibrated, then fused, and finally re-calibrated to obtain a single decision threshold across keywords. While the above techniques are theoretically correct, they rely on meta-parameter tuning and are prone to over-fitting. This study presents an efficient and effective score calibration technique for keyword detection that is based on the logistic regression calibration approach commonly used in forensic speaker identification. The technique applies seamlessly to both single systems and to system fusion, and enables optimization for specific keyword detection evaluation functions. We run experiments on a Vietnamese STD task, comparing the technique with more empirical calibration and fusion schemes and demonstrate that we can achieve comparable or better performance in terms of the NIST ATWV metric with a more elegant solution.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"72 1","pages":"7138-7142"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"74073498","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.2014.6855232
D. Manatakis, Michael G. Nennes, I. Bakas, E. Manolakos
Assessing how the performance of a decentralized wireless sensor network (WSN) algorithm's implementation scales, in terms of communication and energy costs, as the network size increases is an essential requirement before its field deployment. Simulations are commonly used for this purpose, especially for large-scale environmental monitoring applications. However, it is difficult to evaluate energy consumption, processing and memory requirements before the algorithm is really ported to a real WSN platform. We propose a method for emulating the operation of collaborative algorithms in large-scale WSNs by re-using a small number of available real sensor nodes. We demonstrate the potential of the proposed simulation-driven WSN emulation approach by using it to estimate how communication and energy costs scale with the network's size when implementing a collaborative algorithm we developed in [12] for tracking the spatiotemporal evolution of a progressing environmental hazard.
{"title":"Simulation-driven emulation of collaborative algorithms to assess their requirements for a large-scale WSN implementation","authors":"D. Manatakis, Michael G. Nennes, I. Bakas, E. Manolakos","doi":"10.1109/ICASSP.2014.6855232","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855232","url":null,"abstract":"Assessing how the performance of a decentralized wireless sensor network (WSN) algorithm's implementation scales, in terms of communication and energy costs, as the network size increases is an essential requirement before its field deployment. Simulations are commonly used for this purpose, especially for large-scale environmental monitoring applications. However, it is difficult to evaluate energy consumption, processing and memory requirements before the algorithm is really ported to a real WSN platform. We propose a method for emulating the operation of collaborative algorithms in large-scale WSNs by re-using a small number of available real sensor nodes. We demonstrate the potential of the proposed simulation-driven WSN emulation approach by using it to estimate how communication and energy costs scale with the network's size when implementing a collaborative algorithm we developed in [12] for tracking the spatiotemporal evolution of a progressing environmental hazard.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"61 1","pages":"8360-8364"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"74298441","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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