In this letter, an effective acoustic feedback cancellation algorithm is proposed based on the normalized sub-band adaptive filter (NSAF). To improve the confliction between fast convergence rate and low misalignment in the NSAF algorithm, a variable step size is designed to automatically vary according to the update state of the filter. The update state of the filter is adaptively detected via the normalized distance between the long term average and the short term average of the tap-weight vector. Simulation results demonstrate that the proposed algorithm has superior performance in terms of convergence rate and misalignment.

This paper presents an adaptive feedback canceller (AFC) based on a pseudo affine projection (PAP) algorithm that can provide fast and stable adaptation to the time-varying environment. The proposed algorithm utilizes the adaptive linear prediction (LP) to obtain the LP coefficients of input signal model and the inverse gain filter (IGF) to alleviate the effect of compensation gain. As a result, when the input is model as an AR signal, the proposed algorithm satisfies the condition for having an almost unbiased estimatie of the feedback path and then its performance is relatively independent of the gain setting of hearing aids. Simulation results showed that the proposed algorithm is capable of obtaining unbaised feedback path estimates and high speech quality.

Dichotomous coordinate descent (DCD) iterations method has been proposed for adaptive feedback cancellation, which uses a fixed number of iterations and a fixed amplitude range. In this paper, improved DCD algorithms are proposed, which substitute the constant number of iterations and the amplitude range with a variable number of iterations(VI) and/or a variable amplitude range(VA). Thus VI-DCD, VA-DCD and VIA-DCD algorithms are obtained. Computer simulations are used to compare the performance of the proposed algorithms against original DCD algorithm, and simulation results demonstrate that significant improvements are achieved in the convergence speed and accuracy. Another notable conclusion by further simulations is that the proposed algorithms achieve superior performance with a real speech segment as the input.

Howling is very annoying problem to the hearing-aid users and it limits the maximum usable gain of hearing aids. We propose a new feedback cancellation system by inserting a time-varying decorrelation filter in the forward path. We use a second-order all-pass filter with control parameters whose time variation is implemented using a low-frequency modulator. A noticeable reduction of weight-vector misalignment is achievable using our proposed method.