Asynchronous Sampling Rate Conversion of Digital Audio Signal

Abstract

At present, in the digital audio processing sampling rate is respectively 32 kHz, 44.1 kHz, 48 kHz [1]. Because of the different criteria, there is much inconvenience in the process of research. Therefore, the sampling rate converter is a must, between any two kinds of sampling rate. In synchronous sampling rate conversion, you can use decimation and interpolation for sampling rate conversion, but in the asynchronous sampling rate system, due to the different input clock pulse with the output clock pulse, the above method cannot achieve. Therefore we introduce the fractional delay filter sampling rate conversion. This article introduces the principle of the sampling rate conversion and the fractional delay filter based on Farrow structure. At last, we simulate asynchronous sampling rate conversion of audio signal through the MATLAB.