This letter proposes a windowing frequency domain adaptive algorithm, which reuses the filtering error to apply window function in the filter updating symmetrically. By using a proper window function to reduce the negative influence of the spectral leakage, the proposed algorithm can significantly improve the performance of the acoustic echo cancellation for speech signals.

In this letter, we propose a frequency domain active noise control system using the time difference simultaneous perturbation method. This method is an algorithm based on the simultaneous perturbation method which updates the coefficients of the noise control filter only by use of the error signal. The time difference simultaneous perturbation method updates the filter coefficients by using one kind of error signal, while the simultaneous perturbation method updates the filter coefficients by using two kinds of error signal. In the ANC systems, the time difference simultaneous perturbation method is superior because ANC systems cannot obtain two error signals at the same time. When this method is applied to ANC systems, the convergence speed can be increased to a maximum of twice that of the conventional method.

In this paper, we propose a frequency domain active noise control (ANC) system without a secondary path model. The proposed system is based on the frequency domain simultaneous perturbation (FDSP) method we have proposed. In this system, the coefficients of the adaptive filter are updated only by error signals. The conventional ANC system using the filtered-x algorithm becomes unstable due to the error between the secondary path, from secondary source to error sensor, and its model. In contrast, the proposed ANC system has the advantage not to use the model. In this paper, we show the principle of the proposed ANC system, and examine its efficiency through computer simulations.

In this paper, fast algorithms for the CMA (constant modulus algorithm), which is one of the widely used algorithms for blind equalizationi are presented. We propose the FBCMA (frequency domain block CMA) which takes advantage of fast linear convolution in the DFT domain by using the overlap save method. For the FBCMA, a nonlinear error function in the frequency domain is derived using Parseval's relation. Also, an adaptive algorithm in the DFT domain is introduced to adjust the frequency domain filter coefficients. For a block size and filter length of N, the multiplications required for the conventional CMA and proposed FBCMA are on the order of O(N2) and O(N log N), respectively.

In this paper, we propose a new structure of the frequency domain adaptive filter (FDAF). The proposed structure is based on the modified DFT pair which consists of the FIR filters, so that un-delayed output signal can be obtained with stable convergence and without accumulated error which are problems for the conventional FDAFs. The convergence performance of the proposed FDAF is examined through the computer simulations in the adaptive line enhancer (ALE) comparing with the conventional FDAF and the DCT domain adaptive filter. Furthermore, in order to improve the error performance of the FDAF, we propose a composite algorithm which consists of the normalized step size algorithm for fast convergence and the variable step size one for small estimation error. The advantage of the proposed algorithm is also confirmed through simulations in the ALE. Finally, we propose a reduction method of the computational complexity of the proposed FDAF. The proposed method is to utilize a part of the FFT flow-graph, so that the computational complexity is reduced to O(N log N).