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Overview of the EVS codec architecture

 
: Dietz, Martin; Multrus, Markus; Eksler, Vaclav; Malenowski, Vladimir; Norvell, Erik; Pobloth, Harald; Lei, Miao; Zhe, Wang; Laaksonen, Lasse; Vasilache, Adriana; Kamamoto, Yutaka; Kikuiri, Kei; Ragot, Stephane; Faure, Julien; Ehara, Hiroyuki; Rajendran, Vivek; Atti, Venkatraman; Sung, Hosang; Oh, Eunmi; Yuan, Hao; Zhu, Changbao

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Institute of Electrical and Electronics Engineers -IEEE-; IEEE Signal Processing Society:
IEEE 40th IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2015. Proceedings. Vol.8 : 19-24 April 2015, Brisbane, Australia
Piscataway, NJ: IEEE, 2015
ISBN: 978-1-4673-6997-8
ISBN: 978-1-4673-6998-5
S.5698-5702
International Conference on Acoustics, Speech, and Signal Processing (ICASSP) <40, 2015, Brisbane>
Englisch
Konferenzbeitrag
Fraunhofer IIS ()
Standards in der Audiocodierung; audio

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.

: http://publica.fraunhofer.de/dokumente/N-369310.html