Overview of the EVS codec architecture
Martin Dietz, Markus Multrus, Vaclav Eksler, Vladimir Malenovsky, Erik Norvell, Harald Pobloth, Lei Miao, Zhe Wang
Abstract
The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.
BibTeX
@inproceedings{icassp2015_overviewoftheevs,
title = {Overview of the EVS codec architecture},
author = {Martin Dietz and Markus Multrus and Vaclav Eksler and Vladimir Malenovsky and Erik Norvell and Harald Pobloth and Lei Miao and Zhe Wang and Lasse Laaksonen and Adriana Vasilache and Yutaka Kamamoto and Kei Kikuiri and Stéphane Ragot and Julien Faure and Hiroyuki Ehara and Vivek Rajendran and Venkatraman Atti and Hosang Sung and Eunmi Oh and Hao Yuan and Changbao Zhu},
booktitle = {ICASSP 2015},
year = {2015}
}