Adaptive blind channel estimation by least squares smoothing

A least squares smoothing (LSS) approach is presented for the blind estimation of single-input multiple-output (SIMO) finite impulse response systems. By exploiting the isomorphic relation between the input and output subspaces, this geometrical approach identifies the channel from a specially formed least squares smoothing error of the channel output. LSS has the finite sample convergence property, i.e., in the absence of noise, the channel is estimated perfectly with only a finite number of data samples. Referred to as the adaptive least squares smoothing (A-LSS) algorithm, the adaptive implementation has a high convergence rate and low computation cost with no matrix operations. A-LSS is order recursive and is implemented in part using a lattice filter. It has the advantage that when the channel order varies, channel estimates can be obtained without structural change of the implementation. For uncorrelated input sequence, the proposed algorithm performs direct deconvolution as a by-product.

[1]  T. Kailath,et al.  A least-squares approach to blind channel identification , 1995, IEEE Trans. Signal Process..

[2]  Philippe Loubaton,et al.  Prediction error method for second-order blind identification , 1997, IEEE Trans. Signal Process..

[3]  Rolf Johannesson,et al.  Algebraic methods for signal processing and communications coding , 1995 .

[4]  Lang Tong,et al.  Blind identification and equalization of multipath channels , 1992, [Conference Record] SUPERCOMM/ICC '92 Discovering a New World of Communications.

[5]  Lang Tong,et al.  Joint order detection and blind channel estimation by least squares smoothing , 1999, IEEE Trans. Signal Process..

[6]  Dirk T. M. Slock,et al.  Blind fractionally-spaced equalization, perfect-reconstruction filter banks and multichannel linear prediction , 1994, Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing.

[7]  Philippe Loubaton,et al.  Prediction error methods for time-domain blind identification of multichannel FIR filters , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[8]  Hui Liu,et al.  Closed-form blind symbol estimation in digital communications , 1995, IEEE Trans. Signal Process..

[9]  Yingbo Hua,et al.  Fast maximum likelihood for blind identification of multiple FIR channels , 1996, IEEE Trans. Signal Process..

[10]  Constantinos B. Papadias,et al.  Further results on blind identification and equalization of multiple FIR channels , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[11]  Chrysostomos L. Nikias,et al.  EVAM: an eigenvector-based algorithm for multichannel blind deconvolution of input colored signals , 1995, IEEE Trans. Signal Process..

[12]  Sailes K. Sengijpta Fundamentals of Statistical Signal Processing: Estimation Theory , 1995 .

[13]  Arogyaswami Paulraj,et al.  A subspace approach to blind space-time signal processing for wireless communication systems , 1997, IEEE Trans. Signal Process..

[14]  Lang Tong,et al.  Blind channel identification based on second-order statistics: a frequency-domain approach , 1995, IEEE Trans. Inf. Theory.

[15]  Hanoch Lev-Ari,et al.  Modular architectures for adaptive multichannel lattice algorithms , 1983, IEEE Trans. Acoust. Speech Signal Process..

[16]  Eric Moulines,et al.  Subspace methods for the blind identification of multichannel FIR filters , 1995, IEEE Trans. Signal Process..

[17]  S. Haykin,et al.  Adaptive Filter Theory , 1986 .

[18]  David Gesbert,et al.  Robust blind channel identification and equalization based on multi-step predictors , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[19]  M. Morf,et al.  A generalized resultant matrix for polynomial matrices , 1976, 1976 IEEE Conference on Decision and Control including the 15th Symposium on Adaptive Processes.

[20]  Fuyun Ling,et al.  A recursive modified Gram-Schmidt algorithm for least- squares estimation , 1986, IEEE Trans. Acoust. Speech Signal Process..

[21]  T. Kailath,et al.  A state-space approach to adaptive RLS filtering , 1994, IEEE Signal Processing Magazine.

[22]  F. Ling,et al.  A generalized multichannel least squares lattice algorithm based on sequential processing stages , 1984 .

[23]  Richard E. Blahut Algebraic Methods for Signal Processing and Communications Coding , 1991 .