Pub Date : 1999-06-20DOI: 10.1109/SCFT.1999.781477
P. Mermelstein, Y. Qian, K. Zarrinkoub
Code-excited linear prediction coding with generalized pitch prediction (GPP-CELP) requires linear prediction filtering of the stochastic codebook output prior to addition of the adaptive codebook (ACE) component. The ACE component represents a sequence of past reconstructed samples passed through a low-pass filter to reflect the reduced pitch periodicity of the higher speech frequencies. The spectrum of the residual manifests broad peaks leading to significantly narrower distributions in the LPC parameter space. Additionally, the quantization error of the residual may be masked by the significantly greater energy of the ACE component. This work compares the quantization requirements for the information required to represent the time-varying LPC filter of the GPP-CELP coder with that of the classical CELP coder. With non-predictive coding of the LPC information a bit-rate reduction from 20 bits/20 ms to 16 bits/20 ms appears feasible without introducing noticeable degradation due to quantization.
{"title":"LPC quantization requirements for the GPP-CELP coder","authors":"P. Mermelstein, Y. Qian, K. Zarrinkoub","doi":"10.1109/SCFT.1999.781477","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781477","url":null,"abstract":"Code-excited linear prediction coding with generalized pitch prediction (GPP-CELP) requires linear prediction filtering of the stochastic codebook output prior to addition of the adaptive codebook (ACE) component. The ACE component represents a sequence of past reconstructed samples passed through a low-pass filter to reflect the reduced pitch periodicity of the higher speech frequencies. The spectrum of the residual manifests broad peaks leading to significantly narrower distributions in the LPC parameter space. Additionally, the quantization error of the residual may be masked by the significantly greater energy of the ACE component. This work compares the quantization requirements for the information required to represent the time-varying LPC filter of the GPP-CELP coder with that of the classical CELP coder. With non-predictive coding of the LPC information a bit-rate reduction from 20 bits/20 ms to 16 bits/20 ms appears feasible without introducing noticeable degradation due to quantization.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"48 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128131715","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781502
P. Ojala, P. Haavisto, A. Lakaniemi, J. Vainio
This paper presents a novel method to estimate the pitch-lag in a speech codec. The pitch-lag is related to the fundamental frequency of the speech signal and an accurate estimation of this parameter is important for the subjective quality of the synthesised speech. A common problem in speech codecs is that the estimation of the pitch-lag often produces a multiple or a sub-multiple of the true pitch value. When these incorrect pitch-lag values are used in speech synthesis the subjective quality of the speech is degraded. This paper presents an improved method where the estimation of the pitch-lag parameter is biased towards the pitch-lag values of the previous speech segments resulting in a consistent set of consecutive pitch-lag values and a high quality reconstructed signal. The classification of speech into voiced and unvoiced parts is used when tracking the pitch-lag values and adapting the pitch track centered weighting function.
{"title":"A novel pitch-lag search method using adaptive weighting and median filtering","authors":"P. Ojala, P. Haavisto, A. Lakaniemi, J. Vainio","doi":"10.1109/SCFT.1999.781502","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781502","url":null,"abstract":"This paper presents a novel method to estimate the pitch-lag in a speech codec. The pitch-lag is related to the fundamental frequency of the speech signal and an accurate estimation of this parameter is important for the subjective quality of the synthesised speech. A common problem in speech codecs is that the estimation of the pitch-lag often produces a multiple or a sub-multiple of the true pitch value. When these incorrect pitch-lag values are used in speech synthesis the subjective quality of the speech is degraded. This paper presents an improved method where the estimation of the pitch-lag parameter is biased towards the pitch-lag values of the previous speech segments resulting in a consistent set of consecutive pitch-lag values and a high quality reconstructed signal. The classification of speech into voiced and unvoiced parts is used when tracking the pitch-lag values and adapting the pitch track centered weighting function.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115417564","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781492
M. Nishiguchi, A. Inoue, Y. Maeda, J. Matsumoto
MPEG-4 parametric speech coding, harmonic vector excitation coding (HVXC) algorithm, is described. New features of the coder includes a quantizer scheme capable of generating 2.0 and 4.0 kbps scalable bit-streams, where 2.0 kbps decoding is possible using a subset of 4.0 kbps bit-stream. Time scale modification of speech is also possible without changing pitch nor phoneme for fast and slow playback mode. Listening tests show that the proposed coding method at 2.0 kbps provides significantly better quality than that of FS1016 CELP at 4.8 kbps. In October 1998, the HVXC coder was adopted to the Final Draft International Standard (FDIS) of MPEG-4 standardization.
{"title":"Parametric speech coding-HVXC at 2.0-4.0 kbps","authors":"M. Nishiguchi, A. Inoue, Y. Maeda, J. Matsumoto","doi":"10.1109/SCFT.1999.781492","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781492","url":null,"abstract":"MPEG-4 parametric speech coding, harmonic vector excitation coding (HVXC) algorithm, is described. New features of the coder includes a quantizer scheme capable of generating 2.0 and 4.0 kbps scalable bit-streams, where 2.0 kbps decoding is possible using a subset of 4.0 kbps bit-stream. Time scale modification of speech is also possible without changing pitch nor phoneme for fast and slow playback mode. Listening tests show that the proposed coding method at 2.0 kbps provides significantly better quality than that of FS1016 CELP at 4.8 kbps. In October 1998, the HVXC coder was adopted to the Final Draft International Standard (FDIS) of MPEG-4 standardization.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116059051","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781465
J. Schnitzler, J. Eggers, C. Erdmann, P. Vary
This paper describes a wideband (7 kHz) speech coding scheme using code-excited linear prediction (CELP) with mixed time and frequency domain excitation. The proposed frequency domain innovation can be used alternatively or in parallel to a time domain codebook. In addition an improved synthesis filter is used consisting of a signal dependent combination of a forward adaptive and a backward adaptive (FA/BA) structure. An experimental codec operating at 15.5 or 20.0 kbit/s is demonstrated.
{"title":"Wideband speech coding using forward/backward adaptive prediction with mixed time/frequency domain excitation","authors":"J. Schnitzler, J. Eggers, C. Erdmann, P. Vary","doi":"10.1109/SCFT.1999.781465","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781465","url":null,"abstract":"This paper describes a wideband (7 kHz) speech coding scheme using code-excited linear prediction (CELP) with mixed time and frequency domain excitation. The proposed frequency domain innovation can be used alternatively or in parallel to a time domain codebook. In addition an improved synthesis filter is used consisting of a signal dependent combination of a forward adaptive and a backward adaptive (FA/BA) structure. An experimental codec operating at 15.5 or 20.0 kbit/s is demonstrated.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125846323","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781490
D.J. Rahikka, T. Fuja, T. Fazel
The U.S. Government has developed and adopted a new Federal standard vocoder which operates at 2400 bps and is called MELP-mixed excitation linear prediction. This algorithm has quite good voice quality under benign error channel conditions. However, when subjected to high error conditions as may be experienced in vehicular applications, correction techniques may be employed which utilize the underlying inter-frame residual redundancy of the MELP parameters. This paper describes experiments conducted on the MELP algorithm when combined with Viterbi convolutional error decoding, and enhanced with maximum a posteriori techniques which capitalize on the redundancy statistics. Both hard and soft Viterbi decoding situations are investigated.
{"title":"Optimized error correction of MELP speech parameters via maximum a posteriori (MAP) techniques","authors":"D.J. Rahikka, T. Fuja, T. Fazel","doi":"10.1109/SCFT.1999.781490","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781490","url":null,"abstract":"The U.S. Government has developed and adopted a new Federal standard vocoder which operates at 2400 bps and is called MELP-mixed excitation linear prediction. This algorithm has quite good voice quality under benign error channel conditions. However, when subjected to high error conditions as may be experienced in vehicular applications, correction techniques may be employed which utilize the underlying inter-frame residual redundancy of the MELP parameters. This paper describes experiments conducted on the MELP algorithm when combined with Viterbi convolutional error decoding, and enhanced with maximum a posteriori techniques which capitalize on the redundancy statistics. Both hard and soft Viterbi decoding situations are investigated.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125532535","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781494
O. Gottesman, A. Gersho
This paper presents an enhanced waveform interpolative (EWI) speech coder at 4 kbps. The system incorporates novel features such as analysis-by-synthesis (AbS) vector-quantization (VQ) of the dispersion-phase, AbS optimization of the slowly evolving waveform (SEW), a special pitch search for transitions, and switched-predictive analysis-by-synthesis gain VQ. Subjective quality tests indicate that it exceeds that of MPEG-4 at 4 kbps and of G.723.1 at 5.3 kbps, and it is slightly better than that of G.723.1 at 6.3 kbps.
{"title":"Enhanced waveform interpolative coding at 4 kbps","authors":"O. Gottesman, A. Gersho","doi":"10.1109/SCFT.1999.781494","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781494","url":null,"abstract":"This paper presents an enhanced waveform interpolative (EWI) speech coder at 4 kbps. The system incorporates novel features such as analysis-by-synthesis (AbS) vector-quantization (VQ) of the dispersion-phase, AbS optimization of the slowly evolving waveform (SEW), a special pitch search for transitions, and switched-predictive analysis-by-synthesis gain VQ. Subjective quality tests indicate that it exceeds that of MPEG-4 at 4 kbps and of G.723.1 at 5.3 kbps, and it is slightly better than that of G.723.1 at 6.3 kbps.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130937160","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781522
J. Epps, W. Holmes
Telephone speech is typically bandlimited to 4 kHz, resulting in a 'muffled' quality. Coding speech with a bandwidth greater than 4 kHz reduces this distortion, but requires a higher bit rate to avoid other types of distortion. An alternative to coding wider bandwidth speech is to exploit correlations between the 0-4 kHz and 4-8 kHz speech bands to re-synthesize wideband speech from decoded narrowband speech. This paper proposes a new technique for highband spectral envelope prediction, based upon codebook mapping with codebooks split by voicing. An objective comparison with several existing methods reveals that this new technique produces the smallest highband spectral distortion. Combined with a suitable highband excitation synthesis scheme, this envelope prediction scheme produces a significant quality improvement in speech that has been coded using narrowband standards.
{"title":"A new technique for wideband enhancement of coded narrowband speech","authors":"J. Epps, W. Holmes","doi":"10.1109/SCFT.1999.781522","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781522","url":null,"abstract":"Telephone speech is typically bandlimited to 4 kHz, resulting in a 'muffled' quality. Coding speech with a bandwidth greater than 4 kHz reduces this distortion, but requires a higher bit rate to avoid other types of distortion. An alternative to coding wider bandwidth speech is to exploit correlations between the 0-4 kHz and 4-8 kHz speech bands to re-synthesize wideband speech from decoded narrowband speech. This paper proposes a new technique for highband spectral envelope prediction, based upon codebook mapping with codebooks split by voicing. An objective comparison with several existing methods reveals that this new technique produces the smallest highband spectral distortion. Combined with a suitable highband excitation synthesis scheme, this envelope prediction scheme produces a significant quality improvement in speech that has been coded using narrowband standards.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129703371","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781504
S. Ragot, R. Salami, R. Lefebvre
The 11.8 kb/s extension of the G.729 codec, also known as Annex E of the G.729 Recommendation, has been ratified by the ITU-T. This paper describes how the related test sequences have been designed, using the fixed-point C simulation of the codec. The design method is based on the concept of coverage, already used in the design of test sequences for the G.729 codec. Coverage ensures that all possible parameter values are observed in the bitstream, and all portions of the algorithm are executed at least once. Experiments showed that this approach guarantees a satisfying reliability.
{"title":"Design of test sequences for G.729 Annex E","authors":"S. Ragot, R. Salami, R. Lefebvre","doi":"10.1109/SCFT.1999.781504","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781504","url":null,"abstract":"The 11.8 kb/s extension of the G.729 codec, also known as Annex E of the G.729 Recommendation, has been ratified by the ITU-T. This paper describes how the related test sequences have been designed, using the fixed-point C simulation of the codec. The design method is based on the concept of coverage, already used in the design of test sequences for the G.729 codec. Coverage ensures that all possible parameter values are observed in the bitstream, and all portions of the algorithm are executed at least once. Experiments showed that this approach guarantees a satisfying reliability.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129791590","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781476
J. Samuelsson, P. Hedelin
Estimates of optimal performance in terms of spectral distortion (SD) for first order time-recursive spectrum coders are presented. Extensions of high rate theory provides us with the formulas to calculate estimates and also tells us how to design coders with optimal VQ point density. For this purpose, the PDF of the current spectrum parameter vector, given the previous, is needed. This conditional PDF is obtained analytically from a model PDF for pairs of consecutive parameter vectors, based on Gaussian mixtures. The theory gives a lower bound of 16 bits to achieve 1 dB SD. Practical coders must base the adaptive codebook design on quantized previous vectors and experiments suggest that another 2-3 bits is needed to achieve 1 dB SD. Informal subjective tests indicate that transparent quality may be maintained at even lower rates.
给出了一阶时间递归频谱编码器在频谱失真(SD)方面的最优性能估计。高速率理论的扩展为我们提供了估计的计算公式，并告诉我们如何设计具有最佳VQ点密度的编码器。为此，在给定前一种情况下，需要当前频谱参数矢量的PDF。基于高斯混合，从连续参数向量对的模型PDF中解析得到了条件PDF。该理论给出了16位的下限以实现1db SD。实际的编码器必须将自适应码本设计基于量化的先前向量，实验表明需要另外2-3位才能实现1 dB SD。非正式的主观测试表明，透明的质量可能以更低的比率保持。
{"title":"Recursive coding of spectrum parameters","authors":"J. Samuelsson, P. Hedelin","doi":"10.1109/SCFT.1999.781476","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781476","url":null,"abstract":"Estimates of optimal performance in terms of spectral distortion (SD) for first order time-recursive spectrum coders are presented. Extensions of high rate theory provides us with the formulas to calculate estimates and also tells us how to design coders with optimal VQ point density. For this purpose, the PDF of the current spectrum parameter vector, given the previous, is needed. This conditional PDF is obtained analytically from a model PDF for pairs of consecutive parameter vectors, based on Gaussian mixtures. The theory gives a lower bound of 16 bits to achieve 1 dB SD. Practical coders must base the adaptive codebook design on quantized previous vectors and experiments suggest that another 2-3 bits is needed to achieve 1 dB SD. Informal subjective tests indicate that transparent quality may be maintained at even lower rates.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125465843","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 : 1999-06-20DOI: 10.1109/SCFT.1999.781495
T. Eriksson, W. Kleijn
For coders which must produce high speech quality, it is beneficial to have a coding structure which gives zero distortion in the waveform when the quantizer error vanishes (asymptotically perfect reconstruction, APR). It is possible to introduce this property to waveform interpolation (WI) coders by using perfect reconstruction filter banks for analysis and synthesis. Unfortunately, the perfect-reconstruction filter banks are, in general, associated with disadvantages such as oversampling, a loss of physical meaning of the parameters, and increased delay. These disadvantages disappear for the filter bank based on the block DFT transform, but the latter method suffers from energy discontinuities. By using a pre-processor in combination with a block-DFT based WI coder, a coding structure is obtained which maintains the advantages of earlier WI coders and adds the APR property. This new structure is most useful for higher rate WI coders.
{"title":"On waveform-interpolation coding with asymptotically perfect reconstruction","authors":"T. Eriksson, W. Kleijn","doi":"10.1109/SCFT.1999.781495","DOIUrl":"https://doi.org/10.1109/SCFT.1999.781495","url":null,"abstract":"For coders which must produce high speech quality, it is beneficial to have a coding structure which gives zero distortion in the waveform when the quantizer error vanishes (asymptotically perfect reconstruction, APR). It is possible to introduce this property to waveform interpolation (WI) coders by using perfect reconstruction filter banks for analysis and synthesis. Unfortunately, the perfect-reconstruction filter banks are, in general, associated with disadvantages such as oversampling, a loss of physical meaning of the parameters, and increased delay. These disadvantages disappear for the filter bank based on the block DFT transform, but the latter method suffers from energy discontinuities. By using a pre-processor in combination with a block-DFT based WI coder, a coding structure is obtained which maintains the advantages of earlier WI coders and adds the APR property. This new structure is most useful for higher rate WI coders.","PeriodicalId":372569,"journal":{"name":"1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)","volume":"280 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-06-20","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122709938","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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