音質最佳化的網路語音封包排程演算法

Abstract

網路語音通訊的服務品質取決於兩個重要因素，網路延遲與封包漏 失。我們提出一個基於通話品質預測模型的適應性播放排程演算法， 其目地是做為網路延遲與封包漏失間最佳權衡的重要依據。本論文中 將呈現一個音質最佳化的準則以及一個新的播放演算法，將播放緩衝 器設計將轉換成一個音質損害最小化的問題，再尋求出整體延遲與封 包漏失的最佳平衡點。有別於現存的演算法，這一個新演算法可以更 廣泛的適用在基於話務調整和基於封包調整的播放緩衝器機制。實驗 結果顯示，在不同的網路話務情況，新的演算法可以使人耳聆聽音質 達到最佳化的效果。

Packet delay and loss are two essential problems to real-time voice transmission over IP networks. In the proposed system, the playout delay is adaptively adjusted based on a simplified version of the conversational-quality E-model. In this thesis, a perceptually motivated optimization criterion and a practically feasible new algorithm are stated for adaptive playout buffer design. This new buffer design is formulated as an unconstrained optimization problem leading to a better balance between end-to-end delay and packet loss and it can be used in conjunction with per-talkspurt as well as per-packet delay adjustment. We compared the perceived speech quality using the E-model methodology for adaptive playout algorithms with fixed and dynamic setting of the safety factor. Experimental results show that the proposed playout buffer algorithm can achieve the optimum perceived speech quality under various network conditions.