COMPARISON OF INTERPOLATION ALGORITHMS FOR PERFORMING SAMPLE RATE CONVERSION OF AN AUDIO RECORDING

Abstract

There are many algorithms for performing sample rate conversion of an audio recording. These vary widely in both complexity and in audio quality. The simplest (e.g. 'polynomial interpolation algorithms' commonly used in synthesizers) are very CPU-efficient but do not perform any filtering of the sound before downsampling (converting from a higher to a lower sample rate). Frequencies left from the original recording above the frequency limit of the down-sampled recording will result in very undesirable 'aliasing noise' and so it is therefore very important to filter out these high frequencies before downsampling. The ideal filter would be a 'brick filter' that cuts off everything above the new frequency limit and leaves everything below it intact.