Pub Date : 2007-11-01DOI: 10.1109/ICSPC.2007.4728442
J. Mohammed
In this paper, a multi-microphone technique that uses upsampling and IIR-RLS filter to achieve noise reduction in speech degraded by additive noise in the reverberant environment is proposed. One of the main unique features of the proposed technique is that it does not require any statistics about the desired signal. In addition, it does not rely on voice activity detection (VAD), thus avoids the performance degradation due to effects of speech detection errors which are permanent with GSC technique. The good performance of the proposed multi-microphone technique over GSC technique for additive noise reduction has been verified via computer simulations.
{"title":"Multi-Microphone Noise Reduction Technique based on Upsampling and IIR-RLS Filter","authors":"J. Mohammed","doi":"10.1109/ICSPC.2007.4728442","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728442","url":null,"abstract":"In this paper, a multi-microphone technique that uses upsampling and IIR-RLS filter to achieve noise reduction in speech degraded by additive noise in the reverberant environment is proposed. One of the main unique features of the proposed technique is that it does not require any statistics about the desired signal. In addition, it does not rely on voice activity detection (VAD), thus avoids the performance degradation due to effects of speech detection errors which are permanent with GSC technique. The good performance of the proposed multi-microphone technique over GSC technique for additive noise reduction has been verified via computer simulations.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"140 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116197485","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728606
M. Khan, M. H. Wondimagegnehu, T. Shimamura
In this paper we propose a novel non-linear blind adaptive algorithm called the Amplitude Banded Sato (ABSato) algorithm for equalization of communication channels. The ABSato algorithm is derived as a modified version of the Amplitude Banded Least Mean Square (ABLMS) algorithm addressed by Shimamura et al. recently. The capability of nonlinear classification the ABLMS algorithm inherently possesses is kept in the ABSato algorithm, resulting in an improvement of equalization performance. Mean square error as well as bit error rate performances are investigated on simple communication channel models. Observation of simulation results show that the ABSato algorithm provides better performance than the standard Sato algorithm on all the communication channel models.
{"title":"Amplitude Banded Sato Algorithm for Blind Channel Equalization","authors":"M. Khan, M. H. Wondimagegnehu, T. Shimamura","doi":"10.1109/ICSPC.2007.4728606","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728606","url":null,"abstract":"In this paper we propose a novel non-linear blind adaptive algorithm called the Amplitude Banded Sato (ABSato) algorithm for equalization of communication channels. The ABSato algorithm is derived as a modified version of the Amplitude Banded Least Mean Square (ABLMS) algorithm addressed by Shimamura et al. recently. The capability of nonlinear classification the ABLMS algorithm inherently possesses is kept in the ABSato algorithm, resulting in an improvement of equalization performance. Mean square error as well as bit error rate performances are investigated on simple communication channel models. Observation of simulation results show that the ABSato algorithm provides better performance than the standard Sato algorithm on all the communication channel models.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"144 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116442037","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728262
S. Douss, F. Touati, M. Loulou
A design of active mixer intended for the "Mode 1" of ultra wideband multi-band OFDM receivers is presented. It is based on a doubly-balanced Gilbert cell type in which bleeding current sources are added in order to improve linearity and gain. Using an adequate input matching network and optimized device size, it was possible to obtain a good operation of the proposed mixer all over the 3.1-4.8 GHz band with AMS 0.35 ¿m CMOS process parameters. The simulated results show a conversion gain of 12.0 dBm, a noise figure of 7.7 dB and an input IP3 above 0 dBm, when the power consumption is 18 mW under 3V supply voltage.
{"title":"A 3.1-4.8 GHz CMOS Mixer Design using Current Bleeding Technique for UWB MB-OFDM Receivers","authors":"S. Douss, F. Touati, M. Loulou","doi":"10.1109/ICSPC.2007.4728262","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728262","url":null,"abstract":"A design of active mixer intended for the \"Mode 1\" of ultra wideband multi-band OFDM receivers is presented. It is based on a doubly-balanced Gilbert cell type in which bleeding current sources are added in order to improve linearity and gain. Using an adequate input matching network and optimized device size, it was possible to obtain a good operation of the proposed mixer all over the 3.1-4.8 GHz band with AMS 0.35 ¿m CMOS process parameters. The simulated results show a conversion gain of 12.0 dBm, a noise figure of 7.7 dB and an input IP3 above 0 dBm, when the power consumption is 18 mW under 3V supply voltage.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"107 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122949353","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728449
Ying Tian, O. Hammi, S. Boumaiza, F. Ghannouchi
This paper presents a comparative study of the performances of various digital signal components separator topologies for linear amplification using non linear components based transmitters. The topologies studied use different subcomponents to perform the most computationally demanding part of the digital signal components separator that is the calculation of the square root function. Three cases are considered: the square root function is implemented using a commercially available square root mega-function, a custom designed square root function, and a one dimensional look-up table. The measurement results show that all topologies achieve comparable throughput performances. However, the look-up table implementation reduces the computational complexity and improves the achievable accuracy.
{"title":"Design and Optimization of Digital Signal Components Separator of LINC Transmitters Using FPGA Processors","authors":"Ying Tian, O. Hammi, S. Boumaiza, F. Ghannouchi","doi":"10.1109/ICSPC.2007.4728449","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728449","url":null,"abstract":"This paper presents a comparative study of the performances of various digital signal components separator topologies for linear amplification using non linear components based transmitters. The topologies studied use different subcomponents to perform the most computationally demanding part of the digital signal components separator that is the calculation of the square root function. Three cases are considered: the square root function is implemented using a commercially available square root mega-function, a custom designed square root function, and a one dimensional look-up table. The measurement results show that all topologies achieve comparable throughput performances. However, the look-up table implementation reduces the computational complexity and improves the achievable accuracy.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"52 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128350090","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728334
M. Khajehnejad, F. Etezadi, A. Olfat
We present a method for exactly solving the optimization problem for a scenario of discrete rate adaptive modulation. The aim is to maximize the spectral efficiency based on the imperfect channel estimation condition, constant power and average bit error rate(BER) constraints for multilevel quadrature amplitude modulation MQAM over Rayleigh flat fading channels with band-limited feedback signal. Transmission rates are not assigned predetermined values before solving the optimization problem. We show that the general optimization solution entails constellation sizes shouldn't be very large. Our proposed method is an iterative algorithm during which two sets of transmission rates and classification regions for signal to noise ratio (SNR) are updated alternately. The joint solution is a special case of Vector Quantization solution with minimum average cost function. We consider the case where channel is estimated by Pilot Symbol Assisted Modulation (PSAM) and Minimum Mean Square Error (MMSE) criterion.
{"title":"Joint Optimization of Rates and Regions for Optimum Performance of Adaptive Modulation","authors":"M. Khajehnejad, F. Etezadi, A. Olfat","doi":"10.1109/ICSPC.2007.4728334","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728334","url":null,"abstract":"We present a method for exactly solving the optimization problem for a scenario of discrete rate adaptive modulation. The aim is to maximize the spectral efficiency based on the imperfect channel estimation condition, constant power and average bit error rate(BER) constraints for multilevel quadrature amplitude modulation MQAM over Rayleigh flat fading channels with band-limited feedback signal. Transmission rates are not assigned predetermined values before solving the optimization problem. We show that the general optimization solution entails constellation sizes shouldn't be very large. Our proposed method is an iterative algorithm during which two sets of transmission rates and classification regions for signal to noise ratio (SNR) are updated alternately. The joint solution is a special case of Vector Quantization solution with minimum average cost function. We consider the case where channel is estimated by Pilot Symbol Assisted Modulation (PSAM) and Minimum Mean Square Error (MMSE) criterion.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128510145","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728405
M. Maesoumi, M. Masnadi-Shirazi
Partial crosstalk cancellation has been proposed to reduce the online complexity of crosstalk canceller in xDSL systems. Because the crosstalk profile changes over time, there is additional requirement that partial crosstalk cancellation provide a very low pre-processing complexity. Also, a much lower online complexity can be obtained if the multi-user power control and partial crosstalk cancellation problems are solved jointly. Currently, this joint problem is formulated as a constrained optimization problem. However, it suffers from per-tone exhaustive search. This paper presents a solution for this joint problem. The problem is considered as a mixed binary-non-convex problem. Then it is reformulated as a mixed binary-convex problem via a successive linear convex relaxation. Finally it is solved by an efficient branch and bound method. The complexity analysis of our algorithm shows that it provide much lower pre-processing complexity than currently proposed algorithms, allowing it to work efficiently in time-varying crosstalk environment.
{"title":"Efficient Branch and Bound Approach to Joint Multii-User Power Control and Partial Crosstalk Cancellation in xDSL Systems","authors":"M. Maesoumi, M. Masnadi-Shirazi","doi":"10.1109/ICSPC.2007.4728405","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728405","url":null,"abstract":"Partial crosstalk cancellation has been proposed to reduce the online complexity of crosstalk canceller in xDSL systems. Because the crosstalk profile changes over time, there is additional requirement that partial crosstalk cancellation provide a very low pre-processing complexity. Also, a much lower online complexity can be obtained if the multi-user power control and partial crosstalk cancellation problems are solved jointly. Currently, this joint problem is formulated as a constrained optimization problem. However, it suffers from per-tone exhaustive search. This paper presents a solution for this joint problem. The problem is considered as a mixed binary-non-convex problem. Then it is reformulated as a mixed binary-convex problem via a successive linear convex relaxation. Finally it is solved by an efficient branch and bound method. The complexity analysis of our algorithm shows that it provide much lower pre-processing complexity than currently proposed algorithms, allowing it to work efficiently in time-varying crosstalk environment.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128968707","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728579
S. Chaudhry, R. Guha
The authors propose a connection admission control (CAC) scheme and a packet scheduling algorithm for the IEEE 802.16e-2005 standard for fixed and mobile broadband wireless access systems. CAC reserves an adaptive temporal channel bandwidth for mobile subscriber stations based on most recent requests to assure seamless handoff of connections, while the scheduler allocates physical layer slots to user packets based on the corresponding application's data rate and latency characteristics. The effectiveness of the proposed architecture is evaluated though simulations. It is shown that 1) when the system is moderately loaded, the proposed CAC performs better than a fixed guard channel scheme in terms of reducing handoff dropping and new call blocking probabilities; 2) the proposed packet scheduling scheme prioritizes real-time over non real-time traffic in accordance with the quality of service (QoS) parameters of service flows defined in the IEEE standard.
针对固定和移动宽带无线接入系统，提出了一种符合IEEE 802.16e-2005标准的连接允许控制(CAC)方案和分组调度算法。CAC根据最近的请求为移动用户站保留一个自适应的临时信道带宽，以确保连接的无缝切换，而调度程序根据相应应用程序的数据速率和延迟特征为用户数据包分配物理层插槽。通过仿真验证了该体系结构的有效性。结果表明:1)当系统负载适中时，所提出的CAC方案在减少切换丢失和新呼叫阻塞概率方面优于固定保护信道方案;2)根据IEEE标准中定义的业务流的QoS (quality of service，服务质量)参数，提出的分组调度方案对实时流量对非实时流量进行优先级排序。
{"title":"Adaptive Connection Admission Control and Packet Scheduling for QoS Provisioning in Mobile WiMAX","authors":"S. Chaudhry, R. Guha","doi":"10.1109/ICSPC.2007.4728579","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728579","url":null,"abstract":"The authors propose a connection admission control (CAC) scheme and a packet scheduling algorithm for the IEEE 802.16e-2005 standard for fixed and mobile broadband wireless access systems. CAC reserves an adaptive temporal channel bandwidth for mobile subscriber stations based on most recent requests to assure seamless handoff of connections, while the scheduler allocates physical layer slots to user packets based on the corresponding application's data rate and latency characteristics. The effectiveness of the proposed architecture is evaluated though simulations. It is shown that 1) when the system is moderately loaded, the proposed CAC performs better than a fixed guard channel scheme in terms of reducing handoff dropping and new call blocking probabilities; 2) the proposed packet scheduling scheme prioritizes real-time over non real-time traffic in accordance with the quality of service (QoS) parameters of service flows defined in the IEEE standard.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"95 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124639181","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728395
M. Sabeti, M. Sadreddini, J. T. Nezhad
In this paper, the Electroencephalogram (EEG) of twenty schizophrenic patients and twenty age-matched healthy subjects are analyzed for classification purposes. Several features including AR model coefficients, band power and fractal dimension are extracted from EEG signals. This paper proposes a new classification method based on association rule mining. The system we propose consists of a preprocessing phase, a phase for mining the resulted transactional database, and a final phase to improve the resulted association rules. In this case, Fuzzy Accuracy-based Classifier System (F-XCS) is used to improve the resulted fuzzy associative rules for discriminating between healthy and schizophrenic subjects. The experimental results show that the method performs well reaching over 80% in accuracy.
{"title":"EEG Signal Classification using an Association Rule-Based Classifier","authors":"M. Sabeti, M. Sadreddini, J. T. Nezhad","doi":"10.1109/ICSPC.2007.4728395","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728395","url":null,"abstract":"In this paper, the Electroencephalogram (EEG) of twenty schizophrenic patients and twenty age-matched healthy subjects are analyzed for classification purposes. Several features including AR model coefficients, band power and fractal dimension are extracted from EEG signals. This paper proposes a new classification method based on association rule mining. The system we propose consists of a preprocessing phase, a phase for mining the resulted transactional database, and a final phase to improve the resulted association rules. In this case, Fuzzy Accuracy-based Classifier System (F-XCS) is used to improve the resulted fuzzy associative rules for discriminating between healthy and schizophrenic subjects. The experimental results show that the method performs well reaching over 80% in accuracy.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"21 4","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120862678","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728358
S. Jimaa
The concept of using joint error norms adaptive algorithm and the resulting least mean switched error (LMSE) algorithm in the adaptation process of the DFE receiver's structures of asynchronous QPSK (DS-CDMA) is investigated. The performance measure is in-terms of bit error rate (BER) and mean square error (MSE) and for various step-sizes, over AWGN and Gaussian mixture type impulsive channels. The proposed algorithm consists of applying the Least Mean Fourth (LMF) algorithm and switching to the Least Mean Square (LMS) algorithm when the absolute value of error is greater than one. The MSE and BER performances of using the LMSE show that the optimum step-size for the LMF algorithm is 1×10-6 while for the NLMS is kept constant at 0.01. Also the performance of the proposed algorithm was examined against the standard NLMS algorithm and it provides fast convergence rate, lower ground noise floor, and similar BER.
{"title":"On the Use of Joint Error-Norms Adaptive Algorithm in QPSK (DS-CDMA) System","authors":"S. Jimaa","doi":"10.1109/ICSPC.2007.4728358","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728358","url":null,"abstract":"The concept of using joint error norms adaptive algorithm and the resulting least mean switched error (LMSE) algorithm in the adaptation process of the DFE receiver's structures of asynchronous QPSK (DS-CDMA) is investigated. The performance measure is in-terms of bit error rate (BER) and mean square error (MSE) and for various step-sizes, over AWGN and Gaussian mixture type impulsive channels. The proposed algorithm consists of applying the Least Mean Fourth (LMF) algorithm and switching to the Least Mean Square (LMS) algorithm when the absolute value of error is greater than one. The MSE and BER performances of using the LMSE show that the optimum step-size for the LMF algorithm is 1×10-6 while for the NLMS is kept constant at 0.01. Also the performance of the proposed algorithm was examined against the standard NLMS algorithm and it provides fast convergence rate, lower ground noise floor, and similar BER.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121340181","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 : 2007-11-01DOI: 10.1109/ICSPC.2007.4728619
G. Baicher
This paper deals with the optimization of a class of M-channel uniform multirate filter bank. The specific case of a cosine modulated pseudo quadrature mirror filter (QMF) bank is considered for which all the filter bank channels are of equal width on the frequency scale. The design and optimization of the filter bank is based on the use of a single prototype filter. The first stage optimization process is based on perturbing the bandwidth and roll-off factor of a raised cosine prototype filter by using a genetic algorithm (GA) technique. Further improvements are subsequently derived by using a downhill Simplex optimization method.
{"title":"Optimal Design of a Class of M-Channel Uniform Filter Bank using Genetic Algorithms","authors":"G. Baicher","doi":"10.1109/ICSPC.2007.4728619","DOIUrl":"https://doi.org/10.1109/ICSPC.2007.4728619","url":null,"abstract":"This paper deals with the optimization of a class of M-channel uniform multirate filter bank. The specific case of a cosine modulated pseudo quadrature mirror filter (QMF) bank is considered for which all the filter bank channels are of equal width on the frequency scale. The design and optimization of the filter bank is based on the use of a single prototype filter. The first stage optimization process is based on perturbing the bandwidth and roll-off factor of a raised cosine prototype filter by using a genetic algorithm (GA) technique. Further improvements are subsequently derived by using a downhill Simplex optimization method.","PeriodicalId":425397,"journal":{"name":"2007 IEEE International Conference on Signal Processing and Communications","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125441512","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}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：