Pub Date : 2008-05-12DOI: 10.1109/ICASSP.2008.4517963
S. Shirali-Shahreza, M. Shalmani
Steganography is the art of hiding information in a cover media without attracting attention. One of the cover media which can be used for steganography is speech. In this paper, we propose a new speech steganography in wavelet domain. In this method, lifting scheme is used to create perfect reconstruction Int2Int wavelets. The data is hidden in some of the Least Significant Bits (LSB) of detail wavelet coefficients. The LSB bits for hiding are selected with a new adaptive algorithm. This algorithm does not hide information in silent parts, so there is no need for silent detection algorithms. This method has zero error in hiding/unhiding process, while normal wavelet domain LSB has about 0.2 % error in equal hiding capacity. This method is a high capacity steganography method which can hide information up to 20% of the input speech. The Signal-to- Noise Ratio (SNR) and listening tests show that the stegano audio is imperceptible from original audio.
{"title":"High capacity error free wavelet Domain Speech Steganography","authors":"S. Shirali-Shahreza, M. Shalmani","doi":"10.1109/ICASSP.2008.4517963","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517963","url":null,"abstract":"Steganography is the art of hiding information in a cover media without attracting attention. One of the cover media which can be used for steganography is speech. In this paper, we propose a new speech steganography in wavelet domain. In this method, lifting scheme is used to create perfect reconstruction Int2Int wavelets. The data is hidden in some of the Least Significant Bits (LSB) of detail wavelet coefficients. The LSB bits for hiding are selected with a new adaptive algorithm. This algorithm does not hide information in silent parts, so there is no need for silent detection algorithms. This method has zero error in hiding/unhiding process, while normal wavelet domain LSB has about 0.2 % error in equal hiding capacity. This method is a high capacity steganography method which can hide information up to 20% of the input speech. The Signal-to- Noise Ratio (SNR) and listening tests show that the stegano audio is imperceptible from original audio.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126239387","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517553
J. Dmochowski, Zicheng Liu, P. Chou
From an audio perspective, the present state of teleconferencing technology leaves something to be desired; speaker overlap is one of the causes of this inadequate performance. To that end, this paper presents a frequency-domain implementation of convolutive BSS specifically designed for the nature of the teleconferencing environment. In addition to presenting a novel depermutation scheme, this paper presents a least-squares post-processing scheme, which exploits segments during which only a subset of all speakers are active. Experiments with simulated and real data demonstrate the ability of the proposed methods to provide SIRs at or near that of the adaptive noise cancellation (ANC) solution which is obtained under idealistic assumptions that the ANC filters are adapted with one source being on at a time.
{"title":"Blind source separation in a distributed microphone meeting environment for improved teleconferencing","authors":"J. Dmochowski, Zicheng Liu, P. Chou","doi":"10.1109/ICASSP.2008.4517553","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517553","url":null,"abstract":"From an audio perspective, the present state of teleconferencing technology leaves something to be desired; speaker overlap is one of the causes of this inadequate performance. To that end, this paper presents a frequency-domain implementation of convolutive BSS specifically designed for the nature of the teleconferencing environment. In addition to presenting a novel depermutation scheme, this paper presents a least-squares post-processing scheme, which exploits segments during which only a subset of all speakers are active. Experiments with simulated and real data demonstrate the ability of the proposed methods to provide SIRs at or near that of the adaptive noise cancellation (ANC) solution which is obtained under idealistic assumptions that the ANC filters are adapted with one source being on at a time.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126318605","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518330
R. Merched
High Doppler effects resulting from fast time varying dispersive channels represent the most critical impairment to channel equalization techniques in block transmissions. In multicarrier systems, it gives rise to the so-called intercarrier interference (ICI), whose modeling for correct data recovery is paramount. Considering a practical scenario where the designer has no control on the transmitter side, we present a novel turbo equalization scheme based on recent frameworks for the time varying channel parametrization via its derivatives. This includes a fast method for estimating the channel derivatives running on a decision-directed turbo equalization scheme that can be implemented at either symbol or bit level. Unlike recent approaches, the derivatives estimation is adaptive, in the sense that at each turbo estimation it incorporates information on previously estimated parameters.
{"title":"Turbo equalization in high doppler mobile environments: Channel estimation, fast algorithms and adaptive solutions","authors":"R. Merched","doi":"10.1109/ICASSP.2008.4518330","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518330","url":null,"abstract":"High Doppler effects resulting from fast time varying dispersive channels represent the most critical impairment to channel equalization techniques in block transmissions. In multicarrier systems, it gives rise to the so-called intercarrier interference (ICI), whose modeling for correct data recovery is paramount. Considering a practical scenario where the designer has no control on the transmitter side, we present a novel turbo equalization scheme based on recent frameworks for the time varying channel parametrization via its derivatives. This includes a fast method for estimating the channel derivatives running on a decision-directed turbo equalization scheme that can be implemented at either symbol or bit level. Unlike recent approaches, the derivatives estimation is adaptive, in the sense that at each turbo estimation it incorporates information on previously estimated parameters.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126199542","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518092
R. Radhakrishnan, C. Bauer
We present a novel video fingerprinting method based on subspace embedding. The proposed method is particularly robust against frame-rate conversion attacks and geometric attacks among other attacks including compression and spatial scaling. Using a sliding window, we extract fingerprints from a group of subsequent video frames. For the generation of the fingerprints, we first calculate the basis vectors of a coarse representation of this group of frames using a singular value decomposition (SVD). Then, we project the coarse representation of the video frames onto a subset of the basis vectors. Thus, we obtain a subspace representation of the input video frames. Finally, we extract the fingerprint bits by projecting a temporal average of these representations onto pseudorandom basis vectors. Since the subspace is estimated from the input video data itself, any global attack on video such as rotation would result in a corresponding change in estimated basis vectors thereby preserving the subspace representation. We present experimental results on 250 hrs of video to show the robustness and sensitivity of the proposed signature extraction method.
{"title":"Robust video fingerprints based on subspace embedding","authors":"R. Radhakrishnan, C. Bauer","doi":"10.1109/ICASSP.2008.4518092","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518092","url":null,"abstract":"We present a novel video fingerprinting method based on subspace embedding. The proposed method is particularly robust against frame-rate conversion attacks and geometric attacks among other attacks including compression and spatial scaling. Using a sliding window, we extract fingerprints from a group of subsequent video frames. For the generation of the fingerprints, we first calculate the basis vectors of a coarse representation of this group of frames using a singular value decomposition (SVD). Then, we project the coarse representation of the video frames onto a subset of the basis vectors. Thus, we obtain a subspace representation of the input video frames. Finally, we extract the fingerprint bits by projecting a temporal average of these representations onto pseudorandom basis vectors. Since the subspace is estimated from the input video data itself, any global attack on video such as rotation would result in a corresponding change in estimated basis vectors thereby preserving the subspace representation. We present experimental results on 250 hrs of video to show the robustness and sensitivity of the proposed signature extraction method.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128081773","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517856
H. Arora, A. Namboodiri, C. V. Jawahar
Super-resolution reconstruction algorithms assume the availability of exact registration and blur parameters. Inaccurate estimation of these parameters adversely affects the quality of the reconstructed image. However, traditional approaches for image registration are either sensitive to image degradations such as variations in blur, illumination and noise, or are limited in the class of image transformations that can be estimated. We propose an accurate registration algorithm that uses the local phase information, which is robust to the above degradations. We derive the theoretical error rate of the estimates in presence of non-ideal band-pass behavior of the filter and show that the error converges to zero over iterations. We also show the invariance of local phase to a class of blur kernels. Experimental results on images taken under varying conditions clearly demonstrates the robustness of our approach.
{"title":"Robust image registration with illumination, blur and noise variations for super-resolution","authors":"H. Arora, A. Namboodiri, C. V. Jawahar","doi":"10.1109/ICASSP.2008.4517856","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517856","url":null,"abstract":"Super-resolution reconstruction algorithms assume the availability of exact registration and blur parameters. Inaccurate estimation of these parameters adversely affects the quality of the reconstructed image. However, traditional approaches for image registration are either sensitive to image degradations such as variations in blur, illumination and noise, or are limited in the class of image transformations that can be estimated. We propose an accurate registration algorithm that uses the local phase information, which is robust to the above degradations. We derive the theoretical error rate of the estimates in presence of non-ideal band-pass behavior of the filter and show that the error converges to zero over iterations. We also show the invariance of local phase to a class of blur kernels. Experimental results on images taken under varying conditions clearly demonstrates the robustness of our approach.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125713662","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517698
Yan Chen, O. Au
In this paper, we propose a simultaneous rate control and video de-noising algorithm based on rate distortion optimization. According to our previous works, video de-noising can be performed by using rate distortion optimization with a lower bound quantization parameter (QP) constraint, where the lower bound QP is determined by the noise variance. Then, we find that the macroblock level rate control method in H.264 can be seen as an approximate solution of a rate distortion optimization problem with a specified rate distortion function. Based on these two studies, we integrate the video de-noising problem and rate control problem to a rate distortion optimization problem. We show the convexity of the problem and derive the optimal solution. To reduce the complexity, we propose to use a suboptimal solution based on simply thresholding. Some experiments are conducted to demonstrate the efficiency and effectiveness of the proposed method.
{"title":"Simultaneous RD-optimized rate control and video de-noising","authors":"Yan Chen, O. Au","doi":"10.1109/ICASSP.2008.4517698","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517698","url":null,"abstract":"In this paper, we propose a simultaneous rate control and video de-noising algorithm based on rate distortion optimization. According to our previous works, video de-noising can be performed by using rate distortion optimization with a lower bound quantization parameter (QP) constraint, where the lower bound QP is determined by the noise variance. Then, we find that the macroblock level rate control method in H.264 can be seen as an approximate solution of a rate distortion optimization problem with a specified rate distortion function. Based on these two studies, we integrate the video de-noising problem and rate control problem to a rate distortion optimization problem. We show the convexity of the problem and derive the optimal solution. To reduce the complexity, we propose to use a suboptimal solution based on simply thresholding. Some experiments are conducted to demonstrate the efficiency and effectiveness of the proposed method.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121923129","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518324
M. Shabany, Keh-Yih Su, P. Gulak
In this paper, a practical pipelined K-best lattice decoder featuring efficient operation over infinite complex lattices is proposed. This feature is a key element that enables it to operate at a significantly lower complexity than currently reported schemes. The main innovation is a simple means of expanding/visiting the intermediate nodes of the search tree on-demand, rather than exhaustively or approximately, and also directly within the complex-domain framework. In addition, a new distributed sorting scheme is developed to keep track of the best candidates at each search phase; the combined expansion and sorting cores are able to find the K best candidates in just K clock cycles. Its support of unbounded infinite lattice decoding distinguishes our work from previous K-best strategies and also allows its complexity to scale sub-linearly with modulation order. Since the expansion and sorting cores cooperate on a data-driven basis, the architecture is well-suited for a pipelined parallel VLSI implementation of the proposed K-best lattice decoder. Comparative results demonstrating the promising performance, complexity and latency profiles of our proposal are provided in the context of the 4x4 MIMO detection problem.
{"title":"A pipelined scalable high-throughput implementation of a near-ML K-best complex lattice decoder","authors":"M. Shabany, Keh-Yih Su, P. Gulak","doi":"10.1109/ICASSP.2008.4518324","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518324","url":null,"abstract":"In this paper, a practical pipelined K-best lattice decoder featuring efficient operation over infinite complex lattices is proposed. This feature is a key element that enables it to operate at a significantly lower complexity than currently reported schemes. The main innovation is a simple means of expanding/visiting the intermediate nodes of the search tree on-demand, rather than exhaustively or approximately, and also directly within the complex-domain framework. In addition, a new distributed sorting scheme is developed to keep track of the best candidates at each search phase; the combined expansion and sorting cores are able to find the K best candidates in just K clock cycles. Its support of unbounded infinite lattice decoding distinguishes our work from previous K-best strategies and also allows its complexity to scale sub-linearly with modulation order. Since the expansion and sorting cores cooperate on a data-driven basis, the architecture is well-suited for a pipelined parallel VLSI implementation of the proposed K-best lattice decoder. Comparative results demonstrating the promising performance, complexity and latency profiles of our proposal are provided in the context of the 4x4 MIMO detection problem.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121938316","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518729
Sofoklis Kakouros, S. Villette, A. Kondoz
The transmission of voice over IP networks is heavily affected by packet losses. An increasingly popular method to increase the error resilience of these systems is the use of multiple description coding (MDC). However, the MDC techniques commonly used tend to add a significant amount of redundancies, which are not always easy to use optimally. In this paper, we propose a simple vector quantisation scheme to maximise MDC performance, and study several factors affecting its performance under various error conditions. The results show that it is possible to obtain good performance under packet loss conditions, while using only limited amounts of redundancy.
{"title":"Quantisation for Multiple Description Coding for voice over IP","authors":"Sofoklis Kakouros, S. Villette, A. Kondoz","doi":"10.1109/ICASSP.2008.4518729","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518729","url":null,"abstract":"The transmission of voice over IP networks is heavily affected by packet losses. An increasingly popular method to increase the error resilience of these systems is the use of multiple description coding (MDC). However, the MDC techniques commonly used tend to add a significant amount of redundancies, which are not always easy to use optimally. In this paper, we propose a simple vector quantisation scheme to maximise MDC performance, and study several factors affecting its performance under various error conditions. The results show that it is possible to obtain good performance under packet loss conditions, while using only limited amounts of redundancy.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127984546","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518168
Yao Li, P. Djurić
Sequential Monte Carlo (SMC) methods, also referred to as particle filters, have been successfully applied to a variety of highly nonlinear problems such as target tracking with sensor networks. In this paper, we propose the application of a new class of SMC methods named cost-reference particle filters (CRPFs) to target tracking with mobile sensors. CRPF techniques have been shown to be a flexible and robust alternative when there is no knowledge about the probability distributions of the noise in the system. The sensors positioning during tracking is determined by the predicted target's location as obtained by the CRPF. The performance of the method is investigated by simulations and compared to tracking with standard particle filters (SPFs).
{"title":"Target tracking with mobile sensors using cost-reference particle filtering","authors":"Yao Li, P. Djurić","doi":"10.1109/ICASSP.2008.4518168","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518168","url":null,"abstract":"Sequential Monte Carlo (SMC) methods, also referred to as particle filters, have been successfully applied to a variety of highly nonlinear problems such as target tracking with sensor networks. In this paper, we propose the application of a new class of SMC methods named cost-reference particle filters (CRPFs) to target tracking with mobile sensors. CRPF techniques have been shown to be a flexible and robust alternative when there is no knowledge about the probability distributions of the noise in the system. The sensors positioning during tracking is determined by the predicted target's location as obtained by the CRPF. The performance of the method is investigated by simulations and compared to tracking with standard particle filters (SPFs).","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115775648","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517669
A. Alqallaf, A. Tewfik, S. Selleck, Rebecca Johnson
Genetic diseases are characterized by the presence of genetic variations. These variations can be described in the form of copy number. Microarray-based comparative genomic hybridization is a high-resolution technique used to measure copy number variations. However, the observed copy numbers are corrupted by noise, making variations breakpoints hard to detect. In this paper, we provide a framework for the analysis of copy number. The first part of the framework uses an extended version of nonlinear diffusion filter as pre-processing technique to denoise the observed data base. The extension accounts for the nonuniform physical distance between probes. The second part uses estimates the relative frequency of local and global genomic variations across multiple samples to identify statistically and biologically significant variations. For evaluation, we provide copy number variations results using simulated and real data samples. We also validate the predicted copy number variation segments of copy number gain and copy number loss using the experimental molecular tests quantitative polymerase chain reaction and show that our proposed approach is superior to popular commercial software.
{"title":"Framework for the analysis of genetic variations across multiple DNA copy number samples","authors":"A. Alqallaf, A. Tewfik, S. Selleck, Rebecca Johnson","doi":"10.1109/ICASSP.2008.4517669","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517669","url":null,"abstract":"Genetic diseases are characterized by the presence of genetic variations. These variations can be described in the form of copy number. Microarray-based comparative genomic hybridization is a high-resolution technique used to measure copy number variations. However, the observed copy numbers are corrupted by noise, making variations breakpoints hard to detect. In this paper, we provide a framework for the analysis of copy number. The first part of the framework uses an extended version of nonlinear diffusion filter as pre-processing technique to denoise the observed data base. The extension accounts for the nonuniform physical distance between probes. The second part uses estimates the relative frequency of local and global genomic variations across multiple samples to identify statistically and biologically significant variations. For evaluation, we provide copy number variations results using simulated and real data samples. We also validate the predicted copy number variation segments of copy number gain and copy number loss using the experimental molecular tests quantitative polymerase chain reaction and show that our proposed approach is superior to popular commercial software.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131998648","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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