Frequency domain equalizers (FEQs) have been applied extensively in multicarrier systems to enhance transmission rate by reducing transmit redundancy in the form of guard interval. The proposed equalization algorithm is able to remove intersymbol and intercarrier interference (ISI and ICI) incurred by the reduction or the absence of this redundancy by properly exploiting null subcarriers that are inherent in standardized multicarrier systems. Unlike previous proposed schemes, the proposed algorithm does not require additional temporal nor spatial diversity at the receiver to mitigate the channel-induced interferences. Simulation results show that our approach outperforms those of other schemes in terms of bit error rate.
{"title":"Frequency domain equalization for OFDM systems with insufficient guard interval using null subcarriers","authors":"Yin-Ray Huang, C. Fung, K. T. Wong","doi":"10.5281/ZENODO.41393","DOIUrl":"https://doi.org/10.5281/ZENODO.41393","url":null,"abstract":"Frequency domain equalizers (FEQs) have been applied extensively in multicarrier systems to enhance transmission rate by reducing transmit redundancy in the form of guard interval. The proposed equalization algorithm is able to remove intersymbol and intercarrier interference (ISI and ICI) incurred by the reduction or the absence of this redundancy by properly exploiting null subcarriers that are inherent in standardized multicarrier systems. Unlike previous proposed schemes, the proposed algorithm does not require additional temporal nor spatial diversity at the receiver to mitigate the channel-induced interferences. Simulation results show that our approach outperforms those of other schemes in terms of bit error rate.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115152644","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}
For real H.264/AVC codec, the bit allocation in rate control is crucial to coding efficiency. Meanwhile, the rate-distortion-optimization (RDO) based mode-decision technique also affects performance considerably. This paper presents a R-D optimized multi-stage rate control scheme where both bit allocation and the Lagrange multiplier (λMODE) selection are considered. At the first stage, frame layer bit allocation is executed by analyzing frame complexity which is represented by mean-absolute-difference (MAD). Secondly, in order to a select suitable mode to generate reasonable bits, λMODE is further adjusted by using coded and remaining bits information. At the last stage, bits are allocated on macroblock (MB) layer and the computed quantization parameter (QP) is further adjusted. Simulation results verify the performance of the proposed algorithm. Compared with the recommended rate control in H.264/AVC reference software, a gain up to 0.89dB in PSNR is achieved, and the subjective coding quality is also improved.
{"title":"Rate-distortion optimized multi-stage rate control algorithm for H.264/AVC video coding","authors":"Shuijiong Wu, Yiqing Huang, Qin Liu, T. Ikenaga","doi":"10.5281/ZENODO.41466","DOIUrl":"https://doi.org/10.5281/ZENODO.41466","url":null,"abstract":"For real H.264/AVC codec, the bit allocation in rate control is crucial to coding efficiency. Meanwhile, the rate-distortion-optimization (RDO) based mode-decision technique also affects performance considerably. This paper presents a R-D optimized multi-stage rate control scheme where both bit allocation and the Lagrange multiplier (λMODE) selection are considered. At the first stage, frame layer bit allocation is executed by analyzing frame complexity which is represented by mean-absolute-difference (MAD). Secondly, in order to a select suitable mode to generate reasonable bits, λMODE is further adjusted by using coded and remaining bits information. At the last stage, bits are allocated on macroblock (MB) layer and the computed quantization parameter (QP) is further adjusted. Simulation results verify the performance of the proposed algorithm. Compared with the recommended rate control in H.264/AVC reference software, a gain up to 0.89dB in PSNR is achieved, and the subjective coding quality is also improved.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"53 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130494817","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}
This paper concerns the problem of uncorrelated periodic disturbance appearing at the error microphone in single-channel feedforward active noise control (ANC) systems. This disturbance, being uncorrelated with the primary noise, cannot be controlled by standard filtered-x least mean square (FxLMS) algorithm, and increases the residual noise. Furthermore, it significantly degrades the steady-state performance of ANC systems. Here we propose a new method which can simultaneously control both the correlated and uncorrelated noise signals. The proposed method comprises three adaptive filters; 1) the FxLMS algorithm-based ANC filter to cancel the primary noise, 2) a separate FxLMS algorithm-based ANC filter to cancel the uncorrelated disturbance, and 3) the LMS algorithm based supporting adaptive filter to generate appropriate signals for the two ANC filters. Computer simulations are carried out which demonstrate that the proposed method can effectively mitigate the correlated and uncorrelated primary disturbances, and gives significantly improved noise reduction performance. This improved performance is achieved at an increased computational complexity.
{"title":"Mitigating uncorrelated periodic disturbance in narrowband active noise control systems","authors":"M. Akhtar, W. Mitsuhashi","doi":"10.5281/ZENODO.41321","DOIUrl":"https://doi.org/10.5281/ZENODO.41321","url":null,"abstract":"This paper concerns the problem of uncorrelated periodic disturbance appearing at the error microphone in single-channel feedforward active noise control (ANC) systems. This disturbance, being uncorrelated with the primary noise, cannot be controlled by standard filtered-x least mean square (FxLMS) algorithm, and increases the residual noise. Furthermore, it significantly degrades the steady-state performance of ANC systems. Here we propose a new method which can simultaneously control both the correlated and uncorrelated noise signals. The proposed method comprises three adaptive filters; 1) the FxLMS algorithm-based ANC filter to cancel the primary noise, 2) a separate FxLMS algorithm-based ANC filter to cancel the uncorrelated disturbance, and 3) the LMS algorithm based supporting adaptive filter to generate appropriate signals for the two ANC filters. Computer simulations are carried out which demonstrate that the proposed method can effectively mitigate the correlated and uncorrelated primary disturbances, and gives significantly improved noise reduction performance. This improved performance is achieved at an increased computational complexity.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126417333","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}
A single-carrier quasi-orthogonal distributed block space-time coding scheme for a wireless collaborative network with one relaying stage composed of four nodes is proposed. This block quasi-orthogonal design can attain full diversity and maximum rate, and is based on simple amplify-and-forward processing at the relay nodes. A cyclic prefix is introduced into the block transmission to overcome channel delay mismatches in the paths through the four relays. In single-carrier operation, these delays are shown to introduce inter-symbol-interference (ISI) in the destination signal. A linear pre-processing scheme is therefore designed to mitigate this ISI through maximizing signal-to-interference-plus-noise ratio (SINR) at the receiver in combination with iterative equalization. To reduce the computational complexity of this scheme the underlying interference matrices are rendered sparse by the pre-processing operation. Simulation studies confirm the improvement in SINR of the proposed approach as compared to one which only uses rectangular pre-windowing.
{"title":"A single-carrier quasi-orthogonal transmission scheme for asynchronous cooperative relay networks","authors":"M. Hayes, J. Chambers, M. Macleod","doi":"10.5281/ZENODO.41398","DOIUrl":"https://doi.org/10.5281/ZENODO.41398","url":null,"abstract":"A single-carrier quasi-orthogonal distributed block space-time coding scheme for a wireless collaborative network with one relaying stage composed of four nodes is proposed. This block quasi-orthogonal design can attain full diversity and maximum rate, and is based on simple amplify-and-forward processing at the relay nodes. A cyclic prefix is introduced into the block transmission to overcome channel delay mismatches in the paths through the four relays. In single-carrier operation, these delays are shown to introduce inter-symbol-interference (ISI) in the destination signal. A linear pre-processing scheme is therefore designed to mitigate this ISI through maximizing signal-to-interference-plus-noise ratio (SINR) at the receiver in combination with iterative equalization. To reduce the computational complexity of this scheme the underlying interference matrices are rendered sparse by the pre-processing operation. Simulation studies confirm the improvement in SINR of the proposed approach as compared to one which only uses rectangular pre-windowing.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"61 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130314335","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}
One fast inter mode decision algorithm is proposed in this paper. The whole algorithm is divided in to two stages. In the pre-stage, by exploiting spatial and temporal information of encoded macrobocks (MBs), a skip mode early detection scheme is proposed. The homogeneity of current MB is also analyzed to filter out small inter modes in this stage. Secondly, during the block matching stage, a motion feature based inter mode decision scheme is introduced by analyzing the motion vector predictor's accuracy, the block overlapping situation and the smoothness of SAD (sum of absolute difference) value. Moreover, the rate distortion cost is checked in an early stage and we set some constraints to speed up the whole decision flow. Experiments show that our algorithm can achieve a speed-up factor of up to 53.4% with negligible bit increment and quality degradation.
{"title":"Fast inter mode decision algorithm based on macroblock and motion feature analysis for H.264/AVC video coding","authors":"Yiqing Huang, Qin Liu, T. Ikenaga","doi":"10.5281/ZENODO.41655","DOIUrl":"https://doi.org/10.5281/ZENODO.41655","url":null,"abstract":"One fast inter mode decision algorithm is proposed in this paper. The whole algorithm is divided in to two stages. In the pre-stage, by exploiting spatial and temporal information of encoded macrobocks (MBs), a skip mode early detection scheme is proposed. The homogeneity of current MB is also analyzed to filter out small inter modes in this stage. Secondly, during the block matching stage, a motion feature based inter mode decision scheme is introduced by analyzing the motion vector predictor's accuracy, the block overlapping situation and the smoothness of SAD (sum of absolute difference) value. Moreover, the rate distortion cost is checked in an early stage and we set some constraints to speed up the whole decision flow. Experiments show that our algorithm can achieve a speed-up factor of up to 53.4% with negligible bit increment and quality degradation.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"141 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116278758","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}
This paper presents an automatic speaker recognition system for intelligence applications. The system has to provide functionalities for a speaker skimming application in which databases of recorded conversations belonging to an ongoing investigation can be annotated and quickly browsed by an operator. The paper discusses the criticalities introduced by the characteristics of the audio signals under consideration - in particular background noise and channel/coding distortions - as well as the requirements and functionalities of the system under development. It is shown that the performance of state-of-the-art approaches degrades significantly in presence of moderately high background noise. Finally, a novel speaker recognizer based on phonetic features and an ensemble classifier is presented. Results show that the proposed approach improves performance on clean audio, and suggest that it can be employed towards improved real-world robustness.
{"title":"An automatic speaker recognition system for intelligence applications","authors":"E. Marchetto, F. Avanzini, F. Flego","doi":"10.5281/ZENODO.41687","DOIUrl":"https://doi.org/10.5281/ZENODO.41687","url":null,"abstract":"This paper presents an automatic speaker recognition system for intelligence applications. The system has to provide functionalities for a speaker skimming application in which databases of recorded conversations belonging to an ongoing investigation can be annotated and quickly browsed by an operator. The paper discusses the criticalities introduced by the characteristics of the audio signals under consideration - in particular background noise and channel/coding distortions - as well as the requirements and functionalities of the system under development. It is shown that the performance of state-of-the-art approaches degrades significantly in presence of moderately high background noise. Finally, a novel speaker recognizer based on phonetic features and an ensemble classifier is presented. Results show that the proposed approach improves performance on clean audio, and suggest that it can be employed towards improved real-world robustness.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134204522","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 : 2009-11-01DOI: 10.1109/IDT.2009.5404138
A. A. Sallab, H. Fahmy, M. Rashwan
Modern speech recognition applications are heading towards embedded systems and hand-held devices. Distributed Speech Recognition (DSR) system architecture emerged to address this kind of applications. Most of the existing implementations of this system are presented in software fashion, with little consideration to the end product platform in which the system will be deployed. In this paper, an optimized hardware implementation of the front end part of the DSR specified in the basic ETSI Aurora standard ETSI ES 201 108 is presented in FPGA platform prototype, with consideration of migration to structured ASIC in case of mass-production. Main design issues and tips are highlighted. Results are presented in terms of hardware resources utilization, comparison of some basic system components to third party reference designs and compliance to the Aurora standard.
现代语音识别应用正朝着嵌入式系统和手持设备发展。分布式语音识别(DSR)系统架构就是为了解决这类应用而出现的。该系统的大多数现有实现都是以软件方式呈现的，很少考虑将部署系统的最终产品平台。本文在FPGA平台原型上对基本ETSI Aurora标准ETSI ES 201108中规定的DSR前端部分进行了硬件优化实现，并考虑了批量生产时向结构化ASIC的迁移。重点介绍了主要的设计问题和技巧。在硬件资源利用率、一些基本系统组件与第三方参考设计的比较以及对Aurora标准的遵从性方面给出了结果。
{"title":"Hardware implementation of distributed speech recognition system front end","authors":"A. A. Sallab, H. Fahmy, M. Rashwan","doi":"10.1109/IDT.2009.5404138","DOIUrl":"https://doi.org/10.1109/IDT.2009.5404138","url":null,"abstract":"Modern speech recognition applications are heading towards embedded systems and hand-held devices. Distributed Speech Recognition (DSR) system architecture emerged to address this kind of applications. Most of the existing implementations of this system are presented in software fashion, with little consideration to the end product platform in which the system will be deployed. In this paper, an optimized hardware implementation of the front end part of the DSR specified in the basic ETSI Aurora standard ETSI ES 201 108 is presented in FPGA platform prototype, with consideration of migration to structured ASIC in case of mass-production. Main design issues and tips are highlighted. Results are presented in terms of hardware resources utilization, comparison of some basic system components to third party reference designs and compliance to the Aurora standard.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117248798","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}
When multiple correlated data channels are simultaneously processed, multichannel-autoregressive (M-AR) processes may be combined with optimal filtering such as Kalman or H∞ for prediction or estimation from noisy observations. However, the estimation of the M-AR parameters from noisy observations is a key issue to be addressed. Off-line or iterative approaches have been proposed recently, but their computational costs are high or some of them may diverge. Using on-line approaches such as EKF and SPKF is of interest, but the size of the state vector to be estimated is quite high. To reduce this size and the resulting computational costs, we suggest using dual optimal filters. In this paper, we study the relevance of cross-coupled Kalman filters and cross-coupled H∞ filters. The comparative simulation study we carry out shows that our approach corresponds to a compromise between computational cost and performances in terms of pole estimation accuracy.
{"title":"Dual H∞ vs dual Kalman filters for M-AR parameter estimation from noisy observations","authors":"Ali Jamoos, É. Grivel, N. Shakarneh, H. A. Nour","doi":"10.5281/ZENODO.41613","DOIUrl":"https://doi.org/10.5281/ZENODO.41613","url":null,"abstract":"When multiple correlated data channels are simultaneously processed, multichannel-autoregressive (M-AR) processes may be combined with optimal filtering such as Kalman or H∞ for prediction or estimation from noisy observations. However, the estimation of the M-AR parameters from noisy observations is a key issue to be addressed. Off-line or iterative approaches have been proposed recently, but their computational costs are high or some of them may diverge. Using on-line approaches such as EKF and SPKF is of interest, but the size of the state vector to be estimated is quite high. To reduce this size and the resulting computational costs, we suggest using dual optimal filters. In this paper, we study the relevance of cross-coupled Kalman filters and cross-coupled H∞ filters. The comparative simulation study we carry out shows that our approach corresponds to a compromise between computational cost and performances in terms of pole estimation accuracy.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"76 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-08-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129283352","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}
A novel Constraint Adaptive Natural Gradient Algorithm (CANA) is proposed based on the Natural Gradient technique, which is capable of rapidly adjusting the response of an array of sensors to a signal coming from Direction of Interest (DoI) and suppressing noises coming from other directions. Constrained optimization techniques have been extended to enhance the look direction signal in the presence of interference using adaptive natural gradient techniques. Mathematical analysis and realistic MATLAB simulations are developed to confirm the fast convergence of the adaptive weights, and the suitability of algorithm for operations in adverse environments.
{"title":"Constraint Adaptive Natural Gradient Algorithm (CANA) for Adaptive Array Processing","authors":"I. Shah, T. Durrani","doi":"10.5281/ZENODO.41802","DOIUrl":"https://doi.org/10.5281/ZENODO.41802","url":null,"abstract":"A novel Constraint Adaptive Natural Gradient Algorithm (CANA) is proposed based on the Natural Gradient technique, which is capable of rapidly adjusting the response of an array of sensors to a signal coming from Direction of Interest (DoI) and suppressing noises coming from other directions. Constrained optimization techniques have been extended to enhance the look direction signal in the presence of interference using adaptive natural gradient techniques. Mathematical analysis and realistic MATLAB simulations are developed to confirm the fast convergence of the adaptive weights, and the suitability of algorithm for operations in adverse environments.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-08-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129289034","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}
The Dirty Paper Trellis Codes (DPTC) watermarking is a very efficient high capacity scheme. Nevertheless, it suffers of two drawbacks: its security weakness and its CPU computation complexity. In a previous work, we treat those drawbacks by proposing a more secure embedding space and a faster embedding technique. In the current paper, we study the robustness of the scheme with the use of a psychovisual mask and a correcting code. First, we explain how to integrate the psychovisual shaping inside of the DPTC watermarking scheme. Second, we treat of the correcting code integration. Two different masks are used, and robustness comparisons are drawn between the previous DPTC and the two protected psychovisual DPTC. One can observe that for a given SSIM degradation, the robustness results are better for the two protected psychovisual DPTC approaches.
{"title":"Psychovisual rotation-based DPTC watermarking scheme","authors":"M. Chaumont","doi":"10.5281/ZENODO.41410","DOIUrl":"https://doi.org/10.5281/ZENODO.41410","url":null,"abstract":"The Dirty Paper Trellis Codes (DPTC) watermarking is a very efficient high capacity scheme. Nevertheless, it suffers of two drawbacks: its security weakness and its CPU computation complexity. In a previous work, we treat those drawbacks by proposing a more secure embedding space and a faster embedding technique. In the current paper, we study the robustness of the scheme with the use of a psychovisual mask and a correcting code. First, we explain how to integrate the psychovisual shaping inside of the DPTC watermarking scheme. Second, we treat of the correcting code integration. Two different masks are used, and robustness comparisons are drawn between the previous DPTC and the two protected psychovisual DPTC. One can observe that for a given SSIM degradation, the robustness results are better for the two protected psychovisual DPTC approaches.","PeriodicalId":106989,"journal":{"name":"2009 17th European Signal Processing Conference","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-08-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128529357","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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