Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6854696
Zeina Mheich, F. Alberge, P. Duhamel
This paper investigates the maximization of the secrecy-achievable rate region for the Gaussian broadcast channel with confidential message (BCCM) using finite input constellations. The maximization is done jointly over symbol positions and their joint probabilities. The secrecy-achievable rate regions are given for various broadcast strategies which differ in their complexity of implementation. We compare these strategies in terms of improvement in achievable rates and we study the impact of finite input alphabet on the secrecy-achievable rates. It is shown that finite alphabet constraints may change well known results holding in the Gaussian case.
{"title":"The impact of finite-alphabet input on the secrecy-achievable rates for broadcast channel with confidential message","authors":"Zeina Mheich, F. Alberge, P. Duhamel","doi":"10.1109/ICASSP.2014.6854696","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854696","url":null,"abstract":"This paper investigates the maximization of the secrecy-achievable rate region for the Gaussian broadcast channel with confidential message (BCCM) using finite input constellations. The maximization is done jointly over symbol positions and their joint probabilities. The secrecy-achievable rate regions are given for various broadcast strategies which differ in their complexity of implementation. We compare these strategies in terms of improvement in achievable rates and we study the impact of finite input alphabet on the secrecy-achievable rates. It is shown that finite alphabet constraints may change well known results holding in the Gaussian case.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"20 1","pages":"5705-5709"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"72583870","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.2014.6854835
M. H. Maruo, J. Bermudez, L. Resende
Beamformer-assisted acoustic echo cancelers have raised a lot of interest lately. The same performance can be obtained with a reduced length acoustic echo canceler (AEC) as the beamformer (BF) performs spatial cancellation. Structures that jointly optimize the BF and the AEC coefficients are preferred in order to exploit synergies. Analytical models have been already derived for the behavior of the direct form implementation of such systems adapted using the constrained least-mean square (CLMS) algorithm. This work extends the analysis to the popular generalized sidelobe canceler (GSC) structure, while allowing for a positive definite step-size matrix. Analytical models are derived for the mean and mean-square behaviors of the adaptive coefficients. Simulation results are shown to be in excellent agreement with the performance predicted by the theory.
{"title":"Statistical analysis of jointly-optimized GSC implementations of beamformer-assisted acoustic echo cancelers","authors":"M. H. Maruo, J. Bermudez, L. Resende","doi":"10.1109/ICASSP.2014.6854835","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854835","url":null,"abstract":"Beamformer-assisted acoustic echo cancelers have raised a lot of interest lately. The same performance can be obtained with a reduced length acoustic echo canceler (AEC) as the beamformer (BF) performs spatial cancellation. Structures that jointly optimize the BF and the AEC coefficients are preferred in order to exploit synergies. Analytical models have been already derived for the behavior of the direct form implementation of such systems adapted using the constrained least-mean square (CLMS) algorithm. This work extends the analysis to the popular generalized sidelobe canceler (GSC) structure, while allowing for a positive definite step-size matrix. Analytical models are derived for the mean and mean-square behaviors of the adaptive coefficients. Simulation results are shown to be in excellent agreement with the performance predicted by the theory.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"19 1","pages":"6394-6398"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"72726646","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.2014.6853711
A. Sugiyama, Ryoji Miyahara
This paper proposes a new generalized sidelobe canceller with a compact array of microphones suitable for mobile terminals. The output of the fixed beamformer (FBF) is further processed by a newly introduced decorrelation unit which has an auxiliary input signal to improve poor interference suppression of FBF in low frequencies. The output of the decorrelation unit is used as the reference signal for the adaptive blocking matrix and the input for the multi-input canceller. Because low and high frequency components are cancelled or suppressed by the decorrelation unit and FBF, better output-signal quality is obtained. Output signal comparison confirms approximately 12 dB higher interference cancellation by the new beamformer.
{"title":"A new generalized sidelobe canceller with a compact array of microphones suitable for mobile terminals","authors":"A. Sugiyama, Ryoji Miyahara","doi":"10.1109/ICASSP.2014.6853711","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853711","url":null,"abstract":"This paper proposes a new generalized sidelobe canceller with a compact array of microphones suitable for mobile terminals. The output of the fixed beamformer (FBF) is further processed by a newly introduced decorrelation unit which has an auxiliary input signal to improve poor interference suppression of FBF in low frequencies. The output of the decorrelation unit is used as the reference signal for the adaptive blocking matrix and the input for the multi-input canceller. Because low and high frequency components are cancelled or suppressed by the decorrelation unit and FBF, better output-signal quality is obtained. Output signal comparison confirms approximately 12 dB higher interference cancellation by the new beamformer.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"18 1","pages":"820-824"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"74569782","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.2014.6854558
A. Chopra, William Reid, B. Evans
In this article, a novel method of performing subband analysis of digital signals is proposed. Conventional subband decomposition algorithms typically use a binary tree filterbank structure comprised of halfband filters. Due to design limitations of finite length filters, conventional decomposition algorithms typically suffer from interference due to aliasing. While longer halfband filters may reduce aliasing, such filters also increase latency and implementation complexity. Our proposed algorithm uses a novel structure of quadrature mirror filters to ensure aliasing is present outside of the spectral region of interest. Simulation results indicate that, compared to conventional algorithms, the proposed algorithm 1) reduces interference from aliasing by over 30dB, 2) reduces signal processing latency, and 3) reduces implementation complexity.
{"title":"Low complexity subband analysis using quadrature mirror filters","authors":"A. Chopra, William Reid, B. Evans","doi":"10.1109/ICASSP.2014.6854558","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854558","url":null,"abstract":"In this article, a novel method of performing subband analysis of digital signals is proposed. Conventional subband decomposition algorithms typically use a binary tree filterbank structure comprised of halfband filters. Due to design limitations of finite length filters, conventional decomposition algorithms typically suffer from interference due to aliasing. While longer halfband filters may reduce aliasing, such filters also increase latency and implementation complexity. Our proposed algorithm uses a novel structure of quadrature mirror filters to ensure aliasing is present outside of the spectral region of interest. Simulation results indicate that, compared to conventional algorithms, the proposed algorithm 1) reduces interference from aliasing by over 30dB, 2) reduces signal processing latency, and 3) reduces implementation complexity.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"78 1","pages":"5021-5025"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"78628683","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.2014.6854974
Kuan-Yu Chen, Hung-Shin Lee, H. Wang, Berlin Chen, Hsin-Hsi Chen
Since more and more multimedia data associated with spoken documents have been made available to the public, spoken document retrieval (SDR) has become an important research subject in the past two decades. The i-vector based framework has been proposed and introduced to language identification (LID) and speaker recognition (SR) tasks recently. The major contribution of the i-vector framework is to reduce a series of acoustic feature vectors of a speech utterance to a low-dimensional vector representation, and then numbers of well-developed postprocessing techniques (such as probabilistic linear discriminative analysis, PLDA) can be readily and effectively used. However, to our best knowledge, there is no research up to date on applying the i-vector framework for SDR or information retrieval (IR). In this paper, we make a step forward to formulate an i-vector based language modeling (IVLM) framework for SDR. Furthermore, we evaluate the proposed IVLM framework with both inductive and transductive learning strategies. We also exploit multi-levels of index features, including word- and subword-level units, in concert with the proposed framework. The results of SDR experiments conducted on the TDT-2 (Topic Detection and Tracking) collection demonstrate the performance merits of our proposed framework when compared to several existing approaches.
{"title":"I-vector based language modeling for spoken document retrieval","authors":"Kuan-Yu Chen, Hung-Shin Lee, H. Wang, Berlin Chen, Hsin-Hsi Chen","doi":"10.1109/ICASSP.2014.6854974","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854974","url":null,"abstract":"Since more and more multimedia data associated with spoken documents have been made available to the public, spoken document retrieval (SDR) has become an important research subject in the past two decades. The i-vector based framework has been proposed and introduced to language identification (LID) and speaker recognition (SR) tasks recently. The major contribution of the i-vector framework is to reduce a series of acoustic feature vectors of a speech utterance to a low-dimensional vector representation, and then numbers of well-developed postprocessing techniques (such as probabilistic linear discriminative analysis, PLDA) can be readily and effectively used. However, to our best knowledge, there is no research up to date on applying the i-vector framework for SDR or information retrieval (IR). In this paper, we make a step forward to formulate an i-vector based language modeling (IVLM) framework for SDR. Furthermore, we evaluate the proposed IVLM framework with both inductive and transductive learning strategies. We also exploit multi-levels of index features, including word- and subword-level units, in concert with the proposed framework. The results of SDR experiments conducted on the TDT-2 (Topic Detection and Tracking) collection demonstrate the performance merits of our proposed framework when compared to several existing approaches.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"115 1","pages":"7083-7088"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"78722278","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.2014.6853579
Dohyoung Lee, K. Plataniotis
This paper introduces a novel metric for image difference prediction, capable of handling color data. The proposed metric, namely, color difference index based on circular hue, is a full-reference based scheme, which independently processes achromatic and chromatic differences of two input color images. Within the framework, chromatic information is analyzed using two perceptual attributes, hue and chroma information, simulating human visual system mechanism. Unlike conventional approaches where the periodic nature of hue is disregarded, we propose to estimate hue difference by adopting theory of circular statistics. Performance of the proposed solution is validated using benchmark image quality assessment databases. Experimental results indicate the effectiveness of the proposed metric against a wide range of distortions, especially on chromatic distortions, making it better suited for color gamut mapping applications.
{"title":"Towards anovel perceptual color difference metric using circular processing of hue components","authors":"Dohyoung Lee, K. Plataniotis","doi":"10.1109/ICASSP.2014.6853579","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853579","url":null,"abstract":"This paper introduces a novel metric for image difference prediction, capable of handling color data. The proposed metric, namely, color difference index based on circular hue, is a full-reference based scheme, which independently processes achromatic and chromatic differences of two input color images. Within the framework, chromatic information is analyzed using two perceptual attributes, hue and chroma information, simulating human visual system mechanism. Unlike conventional approaches where the periodic nature of hue is disregarded, we propose to estimate hue difference by adopting theory of circular statistics. Performance of the proposed solution is validated using benchmark image quality assessment databases. Experimental results indicate the effectiveness of the proposed metric against a wide range of distortions, especially on chromatic distortions, making it better suited for color gamut mapping applications.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"80 1","pages":"166-170"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"75948291","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.2014.6854576
C. Benedek
In this paper we introduce a probabilistic approach for extracting complex hierarchical object structures from digital images. The proposed framework extends conventional Marked Point Process models by (i) admitting object-subobject ensembles in parent-child relationships and (ii) allowing corresponding objects to form coherent object groups. The proposed method is demonstrated in three application areas: optical circuit inspection, built in area analysis in aerial images, and traffic monitoring on airborne Lidar data.
{"title":"Hierarchical image content analysis with an embedded marked point process framework","authors":"C. Benedek","doi":"10.1109/ICASSP.2014.6854576","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854576","url":null,"abstract":"In this paper we introduce a probabilistic approach for extracting complex hierarchical object structures from digital images. The proposed framework extends conventional Marked Point Process models by (i) admitting object-subobject ensembles in parent-child relationships and (ii) allowing corresponding objects to form coherent object groups. The proposed method is demonstrated in three application areas: optical circuit inspection, built in area analysis in aerial images, and traffic monitoring on airborne Lidar data.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"42 1","pages":"5110-5114"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"76151370","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.2014.6854691
S. Kundu, D. Pados, S. Batalama
We consider arbitrary Hybrid-Automatic-Repeat-Request (H-ARQ) wireless links over quasi-static Rayleigh fading channels. In this paper, we translate the repeat-request advantage of the intended receiver over potential eavesdroppers to link security. In particular, with statistical-only knowledge of the channel and noise, we find for the first time in the literature the optimal power allocation sequence over the H-ARQ rounds that maximizes the outage probability of eavesdroppers for any given target outage probability of the trusted receiver. Simulation studies demonstrate orders of magnitude difference in outage probability between eavesdroppers and intended receiver.
{"title":"Hybrid-ARQ as a communications security measure","authors":"S. Kundu, D. Pados, S. Batalama","doi":"10.1109/ICASSP.2014.6854691","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854691","url":null,"abstract":"We consider arbitrary Hybrid-Automatic-Repeat-Request (H-ARQ) wireless links over quasi-static Rayleigh fading channels. In this paper, we translate the repeat-request advantage of the intended receiver over potential eavesdroppers to link security. In particular, with statistical-only knowledge of the channel and noise, we find for the first time in the literature the optimal power allocation sequence over the H-ARQ rounds that maximizes the outage probability of eavesdroppers for any given target outage probability of the trusted receiver. Simulation studies demonstrate orders of magnitude difference in outage probability between eavesdroppers and intended receiver.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"98 1","pages":"5681-5685"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"75024833","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.2014.6854562
Hung-Yi Cheng, Chun-Yuan Chu, Yen-Liang Chen, A. Wu
The decision feedback equalizer (DFE) is an efficient scheme to suppress intersymbol interference (ISI) in various communication and magnetic recording systems. However, most DFE implementations suffer from the phenomenon of error propagation, which degrades its bit error rate (BER) performance. In this paper, We use sphere detector (SD) to achieve maximum likelihood (ML) detection and significantly reduce the system symbol error rate (SER). Simulations show that the proposed scheme with sphere detector decision feedback equalizer (SD-DFE) algorithm can efficiently reduce the SER. At SNR=28, the SER can be improved from 2.0 × 10-5 (Ideal DFE) to 1.8 × 10-6 (six-stage SD-DFE).
{"title":"Robust decision feedback equalizer scheme by using sphere-decoding detector","authors":"Hung-Yi Cheng, Chun-Yuan Chu, Yen-Liang Chen, A. Wu","doi":"10.1109/ICASSP.2014.6854562","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854562","url":null,"abstract":"The decision feedback equalizer (DFE) is an efficient scheme to suppress intersymbol interference (ISI) in various communication and magnetic recording systems. However, most DFE implementations suffer from the phenomenon of error propagation, which degrades its bit error rate (BER) performance. In this paper, We use sphere detector (SD) to achieve maximum likelihood (ML) detection and significantly reduce the system symbol error rate (SER). Simulations show that the proposed scheme with sphere detector decision feedback equalizer (SD-DFE) algorithm can efficiently reduce the SER. At SNR=28, the SER can be improved from 2.0 × 10-5 (Ideal DFE) to 1.8 × 10-6 (six-stage SD-DFE).","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"2 1","pages":"5041-5044"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"75028791","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.2014.6854407
Sai Han, T. Fingscheidt
Variable-length codes (VLCs) are widely used in media transmission. Compared to fixed-length codes (FLCs), VLCs can represent the same message with a lower bit rate, thus having a better compression performance. But inevitably, VLCs are very sensitive to transmission errors. In this work, based on the trellis representation for VLCs and the BCJR algorithm, we present a variable-length soft-decision decoder utilizing bit-wise channel reliability information and achieving a better error robustness in contrast to hard-decision decoding. Given the application of VLCs in audio coding showing both source correlation and variable block lengths, a strong dependency of performance is observed for both. Therefore, we point out tradeoffs of (soft-decision) decoded FLCs and VLCs depending on quantization bit rate, source correlation, and block length. We find that VLCs over AWGN channels are only recommended for very low source correlation in combination with very short block lengths and soft-decision decoding.
{"title":"Variable-length versus fixed-length coding: On tradeoffs for soft-decision decoding","authors":"Sai Han, T. Fingscheidt","doi":"10.1109/ICASSP.2014.6854407","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854407","url":null,"abstract":"Variable-length codes (VLCs) are widely used in media transmission. Compared to fixed-length codes (FLCs), VLCs can represent the same message with a lower bit rate, thus having a better compression performance. But inevitably, VLCs are very sensitive to transmission errors. In this work, based on the trellis representation for VLCs and the BCJR algorithm, we present a variable-length soft-decision decoder utilizing bit-wise channel reliability information and achieving a better error robustness in contrast to hard-decision decoding. Given the application of VLCs in audio coding showing both source correlation and variable block lengths, a strong dependency of performance is observed for both. Therefore, we point out tradeoffs of (soft-decision) decoded FLCs and VLCs depending on quantization bit rate, source correlation, and block length. We find that VLCs over AWGN channels are only recommended for very low source correlation in combination with very short block lengths and soft-decision decoding.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"141 1","pages":"4269-4273"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"75052692","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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