Research on Adaptive Speech Enhancement Algorithm Based on Differential Microphone Array and Its Implementation with DSP
基于微分麦克风阵列的自适应语音增强算法研究及DSP实现

Adaptive filtering is in common use in most speech enhancement algorithms, while complexity and convergence speed should be considered first when an adaptive filtering method is designed. This paper presents a speech enhancement adaptive filtering method implemented on chip. This method is implemented in two steps. In the first step, first-order differential microphone array is utilized to obtain real-time noise estimation. In the second step, the traditional affine projection algorithm (APA) is modified so that a fast error vector calculation method is obtained, and the search step and projection dimension can be altered dynamically according to the estimation error. Thus, noise can be removed adaptively. This method is realized on a TMS320VC5509 DSP chip. Experiments show that the proposed method has fast convergence speed like RLS (Recursive least squares) and low computational complexity like LMS (Least mean squares).