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|Title:||Optimization of source and channel coding for voice over IP||Authors:||Huang, Y.
|Issue Date:||2005||Citation:||Huang, Y.,Korhonen, J.,Wang, Y. (2005). Optimization of source and channel coding for voice over IP. IEEE International Conference on Multimedia and Expo, ICME 2005 2005 : 173-176. ScholarBank@NUS Repository. https://doi.org/10.1109/ICME.2005.1521388||Abstract:||Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system. © 2005 IEEE.||Source Title:||IEEE International Conference on Multimedia and Expo, ICME 2005||URI:||http://scholarbank.nus.edu.sg/handle/10635/41350||ISBN:||0780393325||DOI:||10.1109/ICME.2005.1521388|
|Appears in Collections:||Staff Publications|
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