Abstract
Perceived voice quality is a key metric in VoIP applications. The quality is mainly affected by IP network impairments such as delay, jitter and packet loss. Playout buffer at the receiving end can be used to compensate for the effects of jitter based on a tradeoff between delay and loss. Adaptive smoothing algorithms are capable of adjusting dynamically the smoothing time based on the network parameters to improve voice quality. In this article, we introduce an efficient and easy perceived quality method for buffer optimization to archive the best voice quality. This work formulates an online loss model which incorporates buffer sizes and applies the Lagrange multiplier approach to optimize the delay-loss problem. Distinct from the other optimal smoothers, the proposed optimal smoother is suitable for any codec and carries the lowest complexity. Simulation experiments validate that the proposed adaptive smoother archives significant improvement in the voice quality.
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© 2005 Springer-Verlag Berlin Heidelberg
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Huang, SF., Wu, E.HK., Chang, PC. (2005). Adaptive Voice Smoother with Optimal Playback Delay for New Generation VoIP Services. In: Yang, L.T., Amamiya, M., Liu, Z., Guo, M., Rammig, F.J. (eds) Embedded and Ubiquitous Computing – EUC 2005. EUC 2005. Lecture Notes in Computer Science, vol 3824. Springer, Berlin, Heidelberg. https://doi.org/10.1007/11596356_100
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DOI: https://doi.org/10.1007/11596356_100
Publisher Name: Springer, Berlin, Heidelberg
Print ISBN: 978-3-540-30807-2
Online ISBN: 978-3-540-32295-5
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