Tele- and video conferencing systems for modern business communication are managed by central hubs, so-called multipoint control units (MCU). One major task of these units is the mixing of audio streams from the participating sites. This is traditionally done by decoding the streams, mixing in time domain and then re-encoding of the mixed signals. This requires additional processing power, leads to increased delay and degraded audio quality. The paper demonstrates how the recently standardized MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec offers a solution to these problems by efficient and delayless mixing in the transform domain of the codec.
Authors:
Albert, Tobias; Ekstrand, Per; Geiger, Ralf; Henn, Fredrik; Lutzky, Manfred; Przioda, Daniel; Ruoppila, Vesa; Schmidt, Markus; Schnell, Markus; Tarnes, Erlend
Affiliations:
Fraunhofer IIS;Coding Technologies;TANDBERG(See document for exact affiliation information.)
AES Convention:
124 (May 2008)
Paper Number:
7337
Publication Date:
May 1, 2008
Subject:
Low Bit-Rate Audio Coding
Click to purchase paper as a non-member or you can login as an AES member to see more options.
No AES members have commented on this paper yet.
To be notified of new comments on this paper you can subscribe to this RSS feed. Forum users should login to see additional options.
If you are not yet an AES member and have something important to say about this paper then we urge you to join the AES today and make your voice heard. You can join online today by clicking here.