Adaptive Playout for VoIP Based on the Enhanced Low Delay AAC Audio Codec

Färber N, Issing J, Lutzky M. Adaptive Playout for VoIP Based on the Enhanced Low Delay AAC Audio Codec. In: AES Convention 124. Audio Engineering Society; 2008. Paper 7395. Available from: https://aes.org/publications/elibrary-page/?id=14525

@inproceedings{Farber2008_14525,
  author = {Färber, Nikolaus and Issing, Jochen and Lutzky, Manfred},
  title = {{Adaptive Playout for VoIP Based on the Enhanced Low Delay AAC Audio Codec}},
  booktitle = {AES Convention 124},
  note = {Paper 7395},
  year = {2008},
  month = may,
  publisher = {Audio Engineering Society},
  url = {https://aes.org/publications/elibrary-page/?id=14525}
}

TY  - CPAPER
TI  - Adaptive Playout for VoIP Based on the Enhanced Low Delay AAC Audio Codec
AU  - Färber, Nikolaus
AU  - Issing, Jochen
AU  - Lutzky, Manfred
T2  - AES Convention 124
M1  - Paper 7395
PY  - 2008
DA  - 2008/05/06
UR  - https://aes.org/publications/elibrary-page/?id=14525
PB  - Audio Engineering Society
LA  - en
AB  - 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.
ER  -