Adaptive voice smoother with optimal playback delay for new generation VoIP services

Research output: Chapter in Book/Report/Conference proceedingConference contributionpeer-review

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.

Original languageEnglish
Title of host publicationEmbedded and Ubiquitous Computing - International Conference EUC 2005, Proceedings
Pages1006-1015
Number of pages10
DOIs
StatePublished - 2005
EventInternational Conference on Embedded and Ubiquitous Computing, EUC 2005 - Nagasaki, Japan
Duration: 6 Dec 20059 Dec 2005

Publication series

NameLecture Notes in Computer Science (including subseries Lecture Notes in Artificial Intelligence and Lecture Notes in Bioinformatics)
Volume3824 LNCS
ISSN (Print)0302-9743
ISSN (Electronic)1611-3349

Conference

ConferenceInternational Conference on Embedded and Ubiquitous Computing, EUC 2005
Country/TerritoryJapan
CityNagasaki
Period6/12/059/12/05

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