Pub Date : 2002-12-01DOI: 10.1109/ICASSP.2001.940481
S. Ohno, G. Giannakis
The adoption of orthogonal frequency-division multiplexing (OFDM) by wireless local area networks and audio/video broadcasting standards testifies to the importance of recovering block precoded transmissions propagating through frequency-selective FIR channels. Existing block transmission standards invoke bandwidth-consuming error control codes to mitigate channel fades and training sequences to identify the FIR channels. To enable low-complexity block-by-block receiver processing, we design redundant precoders with cyclic prefix (CP) and superimposed training sequences for optimal channel estimation and guaranteed symbol recovery regardless of the underlying FIR frequency-selective channels. Numerical results axe presented to access the performance of the designed training and precoding schemes.
{"title":"Optimal training and redundant precoding for block transmissions with application to wireless OFDM","authors":"S. Ohno, G. Giannakis","doi":"10.1109/ICASSP.2001.940481","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940481","url":null,"abstract":"The adoption of orthogonal frequency-division multiplexing (OFDM) by wireless local area networks and audio/video broadcasting standards testifies to the importance of recovering block precoded transmissions propagating through frequency-selective FIR channels. Existing block transmission standards invoke bandwidth-consuming error control codes to mitigate channel fades and training sequences to identify the FIR channels. To enable low-complexity block-by-block receiver processing, we design redundant precoders with cyclic prefix (CP) and superimposed training sequences for optimal channel estimation and guaranteed symbol recovery regardless of the underlying FIR frequency-selective channels. Numerical results axe presented to access the performance of the designed training and precoding schemes.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"125 7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124663981","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940495
R. Heath, H. Bölcskei, A. Paulraj
Wireless systems with multiple transmit and receive antennas (MIMO systems) provide high capacity due to the plurality of modes available in the channel. Previous code designs for MIMO systems have focused primarily on multiplexed signaling for high data rate or diversity signaling for high link reliability. Based on Ganesan and Stoica (2000) and Hassibi and Hochwald (2000), and using results from frame theory, we present a MIMO space-time code design which bridges the gap between multiplexing and diversity and performs well both in terms of ergodic capacity as well as error-probability. In particular, we demonstrate that designs performing well from an ergodic capacity point of view do not necessarily perform well from an error probability point of view. Simulations illustrate performance of the proposed codes in narrowband MIMO Rayleigh fading channels.
{"title":"Space-time signaling and frame theory","authors":"R. Heath, H. Bölcskei, A. Paulraj","doi":"10.1109/ICASSP.2001.940495","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940495","url":null,"abstract":"Wireless systems with multiple transmit and receive antennas (MIMO systems) provide high capacity due to the plurality of modes available in the channel. Previous code designs for MIMO systems have focused primarily on multiplexed signaling for high data rate or diversity signaling for high link reliability. Based on Ganesan and Stoica (2000) and Hassibi and Hochwald (2000), and using results from frame theory, we present a MIMO space-time code design which bridges the gap between multiplexing and diversity and performs well both in terms of ergodic capacity as well as error-probability. In particular, we demonstrate that designs performing well from an ergodic capacity point of view do not necessarily perform well from an error probability point of view. Simulations illustrate performance of the proposed codes in narrowband MIMO Rayleigh fading channels.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"205 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116190990","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940904
H. Wang, H. Meng, Patrick Schone, Berlin Chen, W. Lo
MEI (Mandarin-English Information) is an English-Chinese crosslingual spoken document retrieval (CL-SDR) system developed during the Johns Hopkins University Summer Workshop 2000. We integrate speech recognition, machine translation, and information retrieval technologies to perform CL-SDR. MEI advocates a multi-scale paradigm, where both Chinese words and subwords (characters and syllables) are used in retrieval. The use of subword units can complement the word unit in handling the problems of Chinese word tokenization ambiguity, Chinese homophone ambiguity, and out-of-vocabulary words in audio indexing. This paper focuses on multi-scale audio indexing in MEI. Experiments are based on the Topic Detection and Tracking Corpora (TDT-2 and TDT-3), where we indexed Voice of America Mandarin news broadcasts by speech recognition on both the word and subword scales. We discuss the development of the MEI syllable recognizer, the representations of spoken documents using overlapping subword n-grams and lattice structures. Results show that augmenting words with subwords is beneficial to CL-SDR performance.
MEI (Mandarin-English Information)是在约翰霍普金斯大学2000年夏季研讨会期间开发的一个中英文跨语言口语文档检索系统。我们整合了语音识别、机器翻译和信息检索技术来执行CL-SDR。MEI倡导多尺度范式，检索时既使用汉语词，也使用汉语子词(字符和音节)。子词单元的使用可以对词单元进行补充，以解决音频标引中存在的汉语词分词歧义、汉语同音字歧义和词汇外词等问题。本文主要研究了MEI中的多尺度音频标引。实验基于主题检测和跟踪语料库(TDT-2和TDT-3)，我们在词和子词尺度上对美国之音普通话新闻广播进行了语音识别索引。我们讨论了MEI音节识别器的发展，使用重叠子词n-gram和晶格结构的口语文档表示。结果表明，用子词扩充词有利于提高CL-SDR的性能。
{"title":"Multi-scale-audio indexing for translingual spoken document retrieval","authors":"H. Wang, H. Meng, Patrick Schone, Berlin Chen, W. Lo","doi":"10.1109/ICASSP.2001.940904","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940904","url":null,"abstract":"MEI (Mandarin-English Information) is an English-Chinese crosslingual spoken document retrieval (CL-SDR) system developed during the Johns Hopkins University Summer Workshop 2000. We integrate speech recognition, machine translation, and information retrieval technologies to perform CL-SDR. MEI advocates a multi-scale paradigm, where both Chinese words and subwords (characters and syllables) are used in retrieval. The use of subword units can complement the word unit in handling the problems of Chinese word tokenization ambiguity, Chinese homophone ambiguity, and out-of-vocabulary words in audio indexing. This paper focuses on multi-scale audio indexing in MEI. Experiments are based on the Topic Detection and Tracking Corpora (TDT-2 and TDT-3), where we indexed Voice of America Mandarin news broadcasts by speech recognition on both the word and subword scales. We discuss the development of the MEI syllable recognizer, the representations of spoken documents using overlapping subword n-grams and lattice structures. Results show that augmenting words with subwords is beneficial to CL-SDR performance.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128290755","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940538
R. Nygaard, A. Katsaggelos
We present a time domain signal compression algorithm based on the coding of line segments which are used to approximate the signal. These segments are fitted in a way that is optimal in the rate distortion sense. The approach is applicable to many types of signals, but in this paper we focus on the compression of electrocardiogram (ECG) signals. As opposed to traditional time-domain algorithms, where heuristics are used to extract representative signal samples from the original signal, an optimization algorithm is formulated for sample selection using graph theory, with linear interpolation applied to the reconstruction of the signal. In this paper the algorithm is generalized by using second order polynomial interpolation for the reconstruction of the signal from the extracted signal samples. The polynomials are fitted in a way that guarantees minimum reconstruction error given an upper bound on the number of bits. The method achieves good performance compared both to the case where linear interpolation is used in reconstruction of the signal and to other state-of-the-art ECG coders.
{"title":"Rate distortion optimal signal compression using second order polynomial approximation","authors":"R. Nygaard, A. Katsaggelos","doi":"10.1109/ICASSP.2001.940538","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940538","url":null,"abstract":"We present a time domain signal compression algorithm based on the coding of line segments which are used to approximate the signal. These segments are fitted in a way that is optimal in the rate distortion sense. The approach is applicable to many types of signals, but in this paper we focus on the compression of electrocardiogram (ECG) signals. As opposed to traditional time-domain algorithms, where heuristics are used to extract representative signal samples from the original signal, an optimization algorithm is formulated for sample selection using graph theory, with linear interpolation applied to the reconstruction of the signal. In this paper the algorithm is generalized by using second order polynomial interpolation for the reconstruction of the signal from the extracted signal samples. The polynomials are fitted in a way that guarantees minimum reconstruction error given an upper bound on the number of bits. The method achieves good performance compared both to the case where linear interpolation is used in reconstruction of the signal and to other state-of-the-art ECG coders.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"224 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124462007","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940473
Yuan-Pei Lin, PeiXi Weng, See-May Phoong
There has been great interest in the design of DMT (discrete multitone) transceivers. An M-band DMT transceiver is called block based if the transmitter and the receiver consist of constant matrices. The commonly used DMT systems are mostly block based, e.g., the DFT based system used in transmission over digital subscriber lines. For an FIR channel of order L, it is known that redundancy of length L enables the receiver to cancel ISI completely. Such a scheme allow us to trade bandwidth for ISI cancellation. In block based DMT (BDMT) systems, the redundancy K is typically chosen to be the same as the order of the channel L. We consider BDMT transceiver with redundancy K /spl les/ L. With the reduced redundancy better bandwidth efficiency can be obtained as demonstrated by examples. Furthermore minimum redundancy for BDMT systems are derived and the transceivers are parameterized whenever intersymbol interference (ISI) solutions exist.
{"title":"Block based DMT systems with reduced redundancy","authors":"Yuan-Pei Lin, PeiXi Weng, See-May Phoong","doi":"10.1109/ICASSP.2001.940473","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940473","url":null,"abstract":"There has been great interest in the design of DMT (discrete multitone) transceivers. An M-band DMT transceiver is called block based if the transmitter and the receiver consist of constant matrices. The commonly used DMT systems are mostly block based, e.g., the DFT based system used in transmission over digital subscriber lines. For an FIR channel of order L, it is known that redundancy of length L enables the receiver to cancel ISI completely. Such a scheme allow us to trade bandwidth for ISI cancellation. In block based DMT (BDMT) systems, the redundancy K is typically chosen to be the same as the order of the channel L. We consider BDMT transceiver with redundancy K /spl les/ L. With the reduced redundancy better bandwidth efficiency can be obtained as demonstrated by examples. Furthermore minimum redundancy for BDMT systems are derived and the transceivers are parameterized whenever intersymbol interference (ISI) solutions exist.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"66 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132542057","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940543
Z. Chi, Zhongfeng Wang, K. Parhi
Protecting short frames using turbo coding is a challenging task because of the small interleave size and the need for transmission efficiency. We explore possible trade-off between power consumption (estimated by the average number of iterations) and performance of turbo decoders when short frame turbo codes are used. Three encoding/decoding schemes are proposed to improve performance of turbo decoder in terms of frame/bit error rate, and to increase the data transmission efficiency whether ARQ protocols are performed or not. Specifically, turbo decoding metrics aided short CRC codes are applied to terminated trellis codes, tail-biting encoded trellis codes and CRC embedded trellis codes with a two-fold purpose: to stop the iterative decoding processes and to detect decoding errors at the last iteration. We show that significant coding gains can be achieved by actually increasing the coding rate with negligible increase in power consumption. Performance improvement is demonstrated over both AWGN and Rayleigh flat fading channels.
{"title":"A study on the performance, power consumption tradeoffs of short frame turbo decoder design","authors":"Z. Chi, Zhongfeng Wang, K. Parhi","doi":"10.1109/ICASSP.2001.940543","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940543","url":null,"abstract":"Protecting short frames using turbo coding is a challenging task because of the small interleave size and the need for transmission efficiency. We explore possible trade-off between power consumption (estimated by the average number of iterations) and performance of turbo decoders when short frame turbo codes are used. Three encoding/decoding schemes are proposed to improve performance of turbo decoder in terms of frame/bit error rate, and to increase the data transmission efficiency whether ARQ protocols are performed or not. Specifically, turbo decoding metrics aided short CRC codes are applied to terminated trellis codes, tail-biting encoded trellis codes and CRC embedded trellis codes with a two-fold purpose: to stop the iterative decoding processes and to detect decoding errors at the last iteration. We show that significant coding gains can be achieved by actually increasing the coding rate with negligible increase in power consumption. Performance improvement is demonstrated over both AWGN and Rayleigh flat fading channels.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128328355","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940574
Xueshi Yang, A. Petropulu, J. Pesquet
Traffic flow in high-speed data network systems is often impulsive and long-range dependent. Impulsiveness implies a heavy-tailed marginal distribution, thus lack of finite second-order statistics. Hence, traditional methods for quantifying the long-range dependence of traffic based on its second-order statistics are not applicable. Long-range dependence and self-similarity play an important role in traffic engineering. We have recently shown that the generalized codifference can quantify the dependence structure of impulsive self-similar processes, such as high-speed network traffic. We propose an estimator for the generalized codifference and provide the conditions for it to be asymptotically consistent. We show that these conditions are satisfied for the EAFRP which is a process proposed for modeling high-speed network traffic. We provide simulation results to demonstrate the properties of the proposed estimator, and show how it can be a useful tool in maintaining fairness among users sharing limited network resources.
{"title":"Estimating long-range dependence in impulsive traffic flows","authors":"Xueshi Yang, A. Petropulu, J. Pesquet","doi":"10.1109/ICASSP.2001.940574","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940574","url":null,"abstract":"Traffic flow in high-speed data network systems is often impulsive and long-range dependent. Impulsiveness implies a heavy-tailed marginal distribution, thus lack of finite second-order statistics. Hence, traditional methods for quantifying the long-range dependence of traffic based on its second-order statistics are not applicable. Long-range dependence and self-similarity play an important role in traffic engineering. We have recently shown that the generalized codifference can quantify the dependence structure of impulsive self-similar processes, such as high-speed network traffic. We propose an estimator for the generalized codifference and provide the conditions for it to be asymptotically consistent. We show that these conditions are satisfied for the EAFRP which is a process proposed for modeling high-speed network traffic. We provide simulation results to demonstrate the properties of the proposed estimator, and show how it can be a useful tool in maintaining fairness among users sharing limited network resources.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131339782","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940424
Boris Maricic, Z. Luo, T. Davidson
We formulate the blind equalization of constant modulus (CM) signals as a convex optimization problem. This is done by performing an algebraic transformation on the direct formulation of the equalization problem and then restricting the set of design variables to a subset of the original feasible set. In particular, we express the blind equalization problem as a linear objective function subject to some linear and semidefiniteness constraints. Such semidefinite programs (SDP) can be efficiently solved using interior point methods. Simulations indicate that our method performs better than the standard methods, whilst requiring significantly fewer data samples.
{"title":"Blind equalization of constant modulus signals via restricted convex optimization","authors":"Boris Maricic, Z. Luo, T. Davidson","doi":"10.1109/ICASSP.2001.940424","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940424","url":null,"abstract":"We formulate the blind equalization of constant modulus (CM) signals as a convex optimization problem. This is done by performing an algebraic transformation on the direct formulation of the equalization problem and then restricting the set of design variables to a subset of the original feasible set. In particular, we express the blind equalization problem as a linear objective function subject to some linear and semidefiniteness constraints. Such semidefinite programs (SDP) can be efficiently solved using interior point methods. Simulations indicate that our method performs better than the standard methods, whilst requiring significantly fewer data samples.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132868287","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940510
A. Abdi, M. Kaveh
For the analysis and design of multielement antenna systems in mobile fading channels, we need a model for the space-time cross correlation among the links of the multiple-input multiple-output (MIMO) channel. We propose a general space-time cross correlation function for narrowband Rayleigh fading MIMO channels, where various parameters of interest such as angle spreads at the base station and the user, the distance between the base station and the user, mean directions of the signal arrivals, array configurations, and Doppler spread are all taken into account. The new space-time cross correlation function includes all the relevant parameters of the MIMO narrowband Rayleigh fading channel in a clean compact form, suitable for both simulation and mathematical analysis. It also covers many known correlation models as special cases. We demonstrate the utility of the new space-time correlation model by clarifying the limitations of a widely-accepted correlation model for MIMO fading channels.
{"title":"Space-time correlation modeling of multielement antenna systems in mobile fading channels","authors":"A. Abdi, M. Kaveh","doi":"10.1109/ICASSP.2001.940510","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940510","url":null,"abstract":"For the analysis and design of multielement antenna systems in mobile fading channels, we need a model for the space-time cross correlation among the links of the multiple-input multiple-output (MIMO) channel. We propose a general space-time cross correlation function for narrowband Rayleigh fading MIMO channels, where various parameters of interest such as angle spreads at the base station and the user, the distance between the base station and the user, mean directions of the signal arrivals, array configurations, and Doppler spread are all taken into account. The new space-time cross correlation function includes all the relevant parameters of the MIMO narrowband Rayleigh fading channel in a clean compact form, suitable for both simulation and mathematical analysis. It also covers many known correlation models as special cases. We demonstrate the utility of the new space-time correlation model by clarifying the limitations of a widely-accepted correlation model for MIMO fading channels.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116986769","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 : 2001-09-26DOI: 10.1109/ICASSP.2001.940491
Yan Xin, Zhengdao Wang, G. Giannakis
We present a unified approach to constructing linear space-time (ST) block codes based on unitary constellation-rotating (ST-CR) precoders. We show that with an arbitrary number of M-transmit and N-receive antennas, ST-CR precoders achieve 1 symbol/sec rate and enjoy maximum diversity gain MN over both quasi-static and fast fading channels. We also compare real with complex rotations to delineate the tradeoff between performance and complexity. Based on a simplified decoder, we study diversity and coding gains as well as information-theoretic aspects of the proposed ST-CR scheme. Compared with ST orthogonally designed (ST-OD) codes, ST-CR precoding provides larger coding gain and maximum mutual information. Though ST-OD codes afford simpler decoding, the tradeoff between performance and rate versus complexity favors the ST-CR codes when M, N or the spectral efficiency of the system increase.
{"title":"Space-time diversity systems based on unitary constellation-rotating precoders","authors":"Yan Xin, Zhengdao Wang, G. Giannakis","doi":"10.1109/ICASSP.2001.940491","DOIUrl":"https://doi.org/10.1109/ICASSP.2001.940491","url":null,"abstract":"We present a unified approach to constructing linear space-time (ST) block codes based on unitary constellation-rotating (ST-CR) precoders. We show that with an arbitrary number of M-transmit and N-receive antennas, ST-CR precoders achieve 1 symbol/sec rate and enjoy maximum diversity gain MN over both quasi-static and fast fading channels. We also compare real with complex rotations to delineate the tradeoff between performance and complexity. Based on a simplified decoder, we study diversity and coding gains as well as information-theoretic aspects of the proposed ST-CR scheme. Compared with ST orthogonally designed (ST-OD) codes, ST-CR precoding provides larger coding gain and maximum mutual information. Though ST-OD codes afford simpler decoding, the tradeoff between performance and rate versus complexity favors the ST-CR codes when M, N or the spectral efficiency of the system increase.","PeriodicalId":416477,"journal":{"name":"2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-09-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131199121","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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