Application of least‐squares method to adaptive channel equalizer with frequency sampling filter
Tetsoru Fujii, Hiroshi Harashima, Hiroshi Miyakawa · Electronics and Communications in Japan (Part I Communications) · 1984
Abstract This paper presents a new design method for a frequency sampling adaptive equalizer using the least‐squares method and reports the results obtained in analyzing the equalizer performance. This system employs an adaptive equalizer configuration in which a cyclic code equal to the tap length is used as a test signal with the receive signal and the reference signal each undergoing a discrete Fourier transformation using a frequency sampling filter and applying the least‐squares method in a one‐to‐one relationship to each DFT coefficient. A condition for the implementation of this system is that the test signal be a cyclic code, but this system requires fewer arithmetic operations than any other system employing the least‐squares method. the system presented here employs a high‐speed convergence algorithm in the initial training period and after convergence is accomplished it switches to a gradient method using a frequency sampling adaptive equalizer. In simulation tests with telephone lines it was found that this equalizer converged in about one‐third the time required for the transversal adaptive equalizer.