The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec extends the application area of the Advanced Audio Coding (AAC) family towards high quality conversational services. Through the support of the full audio bandwidth at low delay and low bit rate, it offers excellent support for enhanced VoIP applications. In this paper we provide a brief overview of the AAC-ELD codec and describe how its codec structure can be exploited for IP transport. The overlapping frames and excellent error concealment make it possible to use frame insertion/deletion in order to adjust the playout time to varying network delay. A playout algorithm is proposed which estimates the jitter on the network and adapts the size of the de-jitter buffer in order to minimizes buffering delay and late loss. Considering typical network conditions and the same average delay, it is shown that the playout algorithm can reduce the loss rate by more than one magnitude compared to fixed playout.
Authors:
Färber, Nikolaus; Issing, Jochen; Lutzky, Manfred
Affiliation:
Fraunhofer IIS
AES Convention:
124 (May 2008)
Paper Number:
7395
Publication Date:
May 1, 2008
Session Subject:
Audio Archiving, Storage, Restoration, and Content Management; Audio Networking
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.