Pub Date : 1997-11-20DOI: 10.1109/ACSSC.1997.680518
E. Erkip, B. Aazhang
We compare the performance of different accessing schemes (frequency division, time division and code division) for the uplink in a wireless communication system. We assume an additive white Gaussian noise channel with multipath fading. We consider the case when the receiver can track the channel parameters, but the transmitters cannot. We look at three different measures of performance: the Shannon capacity, the delay limited capacity and probability of outage. Shannon capacity is better suited for systems where the delay requirements are not as stringent, whereas the delay limited capacity and probability of outage give the performance when there are strict delay requirements or when the channel is slowly fading. We observe that all three schemes have the same achievable rates of transmission in Shannon sense, but the code division multiple access scheme performs better than frequency and time division in terms of delay limited capacity and probability of outage.
{"title":"A comparative study of multiple accessing schemes","authors":"E. Erkip, B. Aazhang","doi":"10.1109/ACSSC.1997.680518","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680518","url":null,"abstract":"We compare the performance of different accessing schemes (frequency division, time division and code division) for the uplink in a wireless communication system. We assume an additive white Gaussian noise channel with multipath fading. We consider the case when the receiver can track the channel parameters, but the transmitters cannot. We look at three different measures of performance: the Shannon capacity, the delay limited capacity and probability of outage. Shannon capacity is better suited for systems where the delay requirements are not as stringent, whereas the delay limited capacity and probability of outage give the performance when there are strict delay requirements or when the channel is slowly fading. We observe that all three schemes have the same achievable rates of transmission in Shannon sense, but the code division multiple access scheme performs better than frequency and time division in terms of delay limited capacity and probability of outage.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116295257","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.680557
M. Zoltowski, Der-Feng Tseng, T. Thomas
A blind channel identification scheme for narrowband digital communications with linear modulation is proposed that relies solely on the respective outputs of at least two spatially separated antennas. The proposed method is based on a deterministic relationship between the respective outputs of two FIR filters fed by the same input signal observed by Xu, Liu, Tong, and Kailath (see IEEE Trans. on Signal Processing, p.2982-93, 1995). It also relies on the use of basis functions derived from the Nyquist symbol waveform to characterize each channel's respective impulse response as first proposed by Schell and Smith (see IEEE Milcom-94, p.128-32, 1994). For urban cellular scenarios where the delay spread is on the order of T/sub 0/, where 1/T/sub 0/ is the symbol rate, we show that the continuous-time channel for a given antenna may be well approximated by a linear combination of a small number of time-shifted versions of the Nyquist symbol waveform. The corresponding time shifts may be equi-spaced across the delay spread regardless of the number of actual multipaths and their respective times of arrival. This leads to a critical observation that for a given antenna the same basis coefficient values characterize both the "real" discrete-time channel realized by symbol-spaced sampling starting at t=0 and the virtual discrete-time channel realized by symbol-spaced sampling starting at t=(T/sub 0/)/2. This, in turn, leads to a channel identification scheme requiring two samples per symbol at each antenna that blindly identifies each channel with a relatively small number of symbols in a moderate SNR scenario. A further result is that a bank of small order FIR equalizing filters spanning roughly the delay spread may be computed directly from the basis function coefficient values for each antenna.
{"title":"On the use of basis functions in blind equalization based on deterministic least squares","authors":"M. Zoltowski, Der-Feng Tseng, T. Thomas","doi":"10.1109/ACSSC.1997.680557","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680557","url":null,"abstract":"A blind channel identification scheme for narrowband digital communications with linear modulation is proposed that relies solely on the respective outputs of at least two spatially separated antennas. The proposed method is based on a deterministic relationship between the respective outputs of two FIR filters fed by the same input signal observed by Xu, Liu, Tong, and Kailath (see IEEE Trans. on Signal Processing, p.2982-93, 1995). It also relies on the use of basis functions derived from the Nyquist symbol waveform to characterize each channel's respective impulse response as first proposed by Schell and Smith (see IEEE Milcom-94, p.128-32, 1994). For urban cellular scenarios where the delay spread is on the order of T/sub 0/, where 1/T/sub 0/ is the symbol rate, we show that the continuous-time channel for a given antenna may be well approximated by a linear combination of a small number of time-shifted versions of the Nyquist symbol waveform. The corresponding time shifts may be equi-spaced across the delay spread regardless of the number of actual multipaths and their respective times of arrival. This leads to a critical observation that for a given antenna the same basis coefficient values characterize both the \"real\" discrete-time channel realized by symbol-spaced sampling starting at t=0 and the virtual discrete-time channel realized by symbol-spaced sampling starting at t=(T/sub 0/)/2. This, in turn, leads to a channel identification scheme requiring two samples per symbol at each antenna that blindly identifies each channel with a relatively small number of symbols in a moderate SNR scenario. A further result is that a bank of small order FIR equalizing filters spanning roughly the delay spread may be computed directly from the basis function coefficient values for each antenna.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"68 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121133079","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.680025
A. Madhukumar, A. Premkumar, H. Abut
In this paper we propose an architecture for low bit rate coding of noisy speech. The input noisy speech is decomposed into multi-resolution signal components using the wavelet transform. Iterative Wiener filtering is used at each level of wavelet analysis to enhance the speech. The system model that evolves during enhancement is processed further to get optimal parameters for the quantization. A multistage vector quantizer is used for compression of decomposed speech. The enhanced speech is reconstructed at the receiving end by a VQ decoder and the necessary wavelet reconstruction network. The speech coding rate for the proposed architecture is estimated to be 2.8 kbps.
{"title":"Wavelet quantization of noisy speech using constrained Wiener filtering","authors":"A. Madhukumar, A. Premkumar, H. Abut","doi":"10.1109/ACSSC.1997.680025","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680025","url":null,"abstract":"In this paper we propose an architecture for low bit rate coding of noisy speech. The input noisy speech is decomposed into multi-resolution signal components using the wavelet transform. Iterative Wiener filtering is used at each level of wavelet analysis to enhance the speech. The system model that evolves during enhancement is processed further to get optimal parameters for the quantization. A multistage vector quantizer is used for compression of decomposed speech. The enhanced speech is reconstructed at the receiving end by a VQ decoder and the necessary wavelet reconstruction network. The speech coding rate for the proposed architecture is estimated to be 2.8 kbps.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"58 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127155480","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.680565
D. Kottke, Jong-Kae Fwu, K. Brown
A novel approach for applying hidden Markov models (HMM) to automatic target recognition (ATR) is proposed. The HMM-ATR captures target and background appearance variability by exploiting flexible statistical models. The method utilizes an unsupervised training procedure to estimate the statistical model parameters. Experiments upon a synthetic aperture radar (SAR) database were performed to test robustness over range of target pose, variation in target to background contrast, and mismatches in training and testing conditions. The results are compared against a template matching approach. The HMM captures target appearance variability well and significantly outperforms template matching in both robustness and flexibility.
{"title":"Hidden Markov modeling for automatic target recognition","authors":"D. Kottke, Jong-Kae Fwu, K. Brown","doi":"10.1109/ACSSC.1997.680565","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680565","url":null,"abstract":"A novel approach for applying hidden Markov models (HMM) to automatic target recognition (ATR) is proposed. The HMM-ATR captures target and background appearance variability by exploiting flexible statistical models. The method utilizes an unsupervised training procedure to estimate the statistical model parameters. Experiments upon a synthetic aperture radar (SAR) database were performed to test robustness over range of target pose, variation in target to background contrast, and mismatches in training and testing conditions. The results are compared against a template matching approach. The HMM captures target appearance variability well and significantly outperforms template matching in both robustness and flexibility.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"47 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123247471","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.679104
W. Lu
This paper describes a new method for the parameterization of compactly supported orthogonal wavelet filters. The well-known Daubechies (1988) orthogonal wavelets can be viewed as a subset in the parameterized orthogonal wavelet class, which processes a maximum number of vanishing moments for a given filter length. Unlike the existing parameterizations of orthogonal wavelets, the proposed method does the parameterization through a linear characterization of all halfband filters. The paper also includes examples of optimal designs of orthogonal wavelets obtained using this parameterization technique in conjunction with efficient linear programming or quadratic programming, and application of these wavelets to signal compression and signal denoising.
{"title":"Linear parameterization of orthogonal wavelets","authors":"W. Lu","doi":"10.1109/ACSSC.1997.679104","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679104","url":null,"abstract":"This paper describes a new method for the parameterization of compactly supported orthogonal wavelet filters. The well-known Daubechies (1988) orthogonal wavelets can be viewed as a subset in the parameterized orthogonal wavelet class, which processes a maximum number of vanishing moments for a given filter length. Unlike the existing parameterizations of orthogonal wavelets, the proposed method does the parameterization through a linear characterization of all halfband filters. The paper also includes examples of optimal designs of orthogonal wavelets obtained using this parameterization technique in conjunction with efficient linear programming or quadratic programming, and application of these wavelets to signal compression and signal denoising.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125549429","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.679087
M. Ercegovac, T. Lang
We describe a redundant radix-2 representation with digit set {0,1,2,3} and an encoding using three bits per digit, instead of the minimum of two. This representation is then used to implement several adders, having different number of redundant and conventional operands. We show that the resulting adders are faster than those using carry-save representation. The evaluations are done for two libraries of standard cells. These adders have applications where redundant adders (with limited carry propagation) are used. This includes sequential and combinational accumulators and multipliers, CORDIC units, and digit-recurrences for operations such as division and square root. We also evaluate the effect of the proposed adders on the delay and size of a 54-bit tree multiplier.
{"title":"Effective coding for fast redundant adders using the radix-2 digit set {0,1,2,3}","authors":"M. Ercegovac, T. Lang","doi":"10.1109/ACSSC.1997.679087","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679087","url":null,"abstract":"We describe a redundant radix-2 representation with digit set {0,1,2,3} and an encoding using three bits per digit, instead of the minimum of two. This representation is then used to implement several adders, having different number of redundant and conventional operands. We show that the resulting adders are faster than those using carry-save representation. The evaluations are done for two libraries of standard cells. These adders have applications where redundant adders (with limited carry propagation) are used. This includes sequential and combinational accumulators and multipliers, CORDIC units, and digit-recurrences for operations such as division and square root. We also evaluate the effect of the proposed adders on the delay and size of a 54-bit tree multiplier.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"177 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123020802","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.679120
S. Kraut, L. T. McWhorter, L. Scharf
We present a unified derivation of the distributions for adaptive versions of matched subspace detectors (MSDs) derived by Scharf (see Statistical Signal Processing, Addison-Wesley, and IEEE Trans. Signal Processing, 1996). These include: (1) the matched filter detector, (2) the gain invariant (CFAR) matched filter detector (3) the phase invariant matched subspace detector, and (4) the gain invariant (CFAR) and phase invariant matched subspace detector. We show that all these detectors can be decomposed into representations that are simple functions of the same five statistically independent, chi-squared or normal, scalar random variables. This canonical representation has at least three advantages: (1) the behavior of these detectors can easily be related to that of the non-adaptive detectors from which they are derived (2) moments can be simply obtained from the distributions of the scalar random variables, and (3) Monte Carlo simulations of the distributions can be implemented more efficiently.
{"title":"A canonical representation for distributions of adaptive matched subspace detectors","authors":"S. Kraut, L. T. McWhorter, L. Scharf","doi":"10.1109/ACSSC.1997.679120","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679120","url":null,"abstract":"We present a unified derivation of the distributions for adaptive versions of matched subspace detectors (MSDs) derived by Scharf (see Statistical Signal Processing, Addison-Wesley, and IEEE Trans. Signal Processing, 1996). These include: (1) the matched filter detector, (2) the gain invariant (CFAR) matched filter detector (3) the phase invariant matched subspace detector, and (4) the gain invariant (CFAR) and phase invariant matched subspace detector. We show that all these detectors can be decomposed into representations that are simple functions of the same five statistically independent, chi-squared or normal, scalar random variables. This canonical representation has at least three advantages: (1) the behavior of these detectors can easily be related to that of the non-adaptive detectors from which they are derived (2) moments can be simply obtained from the distributions of the scalar random variables, and (3) Monte Carlo simulations of the distributions can be implemented more efficiently.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114297946","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.680549
M. Viberg, H. Krim
Array signal processing is the common name for a large number of signal processing techniques involving parameter estimation from multichannel data. The prototype problem is to find the directions of incoming wavefronts using an antenna array. The applications are numerous, including unexpected problems not involving spatially distributed sensors. A vast number of estimation methods have been proposed and extensively analyzed over the last 2-3 decades. The paper provides an introduction to the various algorithms.
{"title":"Two decades of statistical array processing","authors":"M. Viberg, H. Krim","doi":"10.1109/ACSSC.1997.680549","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680549","url":null,"abstract":"Array signal processing is the common name for a large number of signal processing techniques involving parameter estimation from multichannel data. The prototype problem is to find the directions of incoming wavefronts using an antenna array. The applications are numerous, including unexpected problems not involving spatially distributed sensors. A vast number of estimation methods have been proposed and extensively analyzed over the last 2-3 decades. The paper provides an introduction to the various algorithms.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129169054","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.680535
W. Lu, A. Antoniou, S. Saab
A method for the design of FIR digital filters with low power consumption is proposed. In this method, the digital filter is implemented as a cascade arrangement of low-order sections. The first section is designed through optimization so as to satisfy as far as possible, the overall required specifications. The first section is then fixed and a second section is added, which is designed so that the first two sections in cascade satisfy again as far as possible the overall required specifications. This process is repeated until a multisection filter is obtained that would satisfy the required specifications under the most critical circumstances imposed by the application at hand. In multisection filters of this type, the minimum number of sections required to process the current input signal can be switched in through the use of a simple adaptation mechanism and, in this way, the power consumption can be minimized. This design strategy is achieved by formulating the design of the k-th section as a weighted least-squares minimization problem, assuming that an optimum (k-1)-section design is available.
{"title":"Sequential design of FIR digital filters for low-power DSP applications","authors":"W. Lu, A. Antoniou, S. Saab","doi":"10.1109/ACSSC.1997.680535","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.680535","url":null,"abstract":"A method for the design of FIR digital filters with low power consumption is proposed. In this method, the digital filter is implemented as a cascade arrangement of low-order sections. The first section is designed through optimization so as to satisfy as far as possible, the overall required specifications. The first section is then fixed and a second section is added, which is designed so that the first two sections in cascade satisfy again as far as possible the overall required specifications. This process is repeated until a multisection filter is obtained that would satisfy the required specifications under the most critical circumstances imposed by the application at hand. In multisection filters of this type, the minimum number of sections required to process the current input signal can be switched in through the use of a simple adaptation mechanism and, in this way, the power consumption can be minimized. This design strategy is achieved by formulating the design of the k-th section as a weighted least-squares minimization problem, assuming that an optimum (k-1)-section design is available.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121207405","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 : 1997-11-02DOI: 10.1109/ACSSC.1997.679069
J. Samuelsson, J. Skoglund, J. Linden
Taking the evolution of spectral parameters into consideration in speech coding has been shown to enhance the perceptual performance. In this study we examine and compare two methods that are designed for explicit control of spectral dynamics. One method operates on the encoder part of the coding system by incorporating a constraint in the distortion measure and the other method smoothes the trajectory of output vectors at the decoder side. The decoder method requires however an additional coding delay of one frame. By means of listening experiments it is demonstrated for three different vector quantizer structures that especially the decoder method gives significant improvements. For noisy channels, the preference for this method is even more emphasized.
{"title":"Controlling spectral dynamics in LPC quantization for perceptual enhancement","authors":"J. Samuelsson, J. Skoglund, J. Linden","doi":"10.1109/ACSSC.1997.679069","DOIUrl":"https://doi.org/10.1109/ACSSC.1997.679069","url":null,"abstract":"Taking the evolution of spectral parameters into consideration in speech coding has been shown to enhance the perceptual performance. In this study we examine and compare two methods that are designed for explicit control of spectral dynamics. One method operates on the encoder part of the coding system by incorporating a constraint in the distortion measure and the other method smoothes the trajectory of output vectors at the decoder side. The decoder method requires however an additional coding delay of one frame. By means of listening experiments it is demonstrated for three different vector quantizer structures that especially the decoder method gives significant improvements. For noisy channels, the preference for this method is even more emphasized.","PeriodicalId":240431,"journal":{"name":"Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers (Cat. No.97CB36136)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114534942","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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