A New Cosine Series Antialiasing Function and its Application to Aliasing-Free Glottal Source Models for Speech and Singing Synthesis

Hideki Kawahara, Ken-Ichi Sakakibara, Masanori Morise, Hideki Banno, Tomoki Toda, Toshio Irino


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.


 DOI: 10.21437/Interspeech.2017-15

Cite as: Kawahara, H., Sakakibara, K., Morise, M., Banno, H., Toda, T., Irino, T. (2017) A New Cosine Series Antialiasing Function and its Application to Aliasing-Free Glottal Source Models for Speech and Singing Synthesis. Proc. Interspeech 2017, 1358-1362, DOI: 10.21437/Interspeech.2017-15.


@inproceedings{Kawahara2017,
  author={Hideki Kawahara and Ken-Ichi Sakakibara and Masanori Morise and Hideki Banno and Tomoki Toda and Toshio Irino},
  title={A New Cosine Series Antialiasing Function and its Application to Aliasing-Free Glottal Source Models for Speech and Singing Synthesis},
  year=2017,
  booktitle={Proc. Interspeech 2017},
  pages={1358--1362},
  doi={10.21437/Interspeech.2017-15},
  url={http://dx.doi.org/10.21437/Interspeech.2017-15}
}