# A new cosine series antialiasing function and its application to aliasing-free glottal source models for speech and singing synthesis

## TL;DR

This paper introduces a novel cosine-series-based antialiasing filter for generating aliasing-free excitation signals, enhancing speech and singing synthesis models with improved accuracy and flexibility.

## Contribution

It presents a new antialiasing filter design procedure and applies it to improve existing speech source models, with open-source MATLAB implementations.

## Key findings

- Successfully generated aliasing-free excitation signals
- Enhanced the reliability of speech synthesis models
- Provided open-source tools for speech science

## Abstract

We formulated and implemented a procedure to generate aliasing-free excitation source signals. It uses a new antialiasing filter in the continuous time domain followed by an IIR digital filter for response equalization. We introduced a cosine-series-based general design procedure for the new antialiasing function. We applied this new procedure to implement the antialiased Fujisaki-Ljungqvist model. We also applied it to revise our previous implementation of the antialiased Fant-Liljencrants model. A combination of these signals and a lattice implementation of the time varying vocal tract model provides a reliable and flexible basis to test fo extractors and source aperiodicity analysis methods. MATLAB implementations of these antialiased excitation source models are available as part of our open source tools for speech science.