網路聲訊標準 G.723.1 編解碼優化之研究

Abstract

G723.1 是一非常重要的音訊標準。目前重要的音視訊會議(video conference)標準,如 H.323, H.324 均採用其為音訊標準。G723.1 之所 以如此受重視的原因如下: 1.其傳輸率非常低,只有5.3 kbps 或 6.3 kbps 。 2.其總延遲(delay)非常短,只有 37.5 msec。 3.同時可維持 較佳的音訊品質。在一項測試中,它得到3.98分(完美電話音訊 品質是 4.00 分) 4. 偶爾封包(packet)之遺失,對音訊品質影響不大基於上述 G.723.1 的優點及重要性,本論文最主要的目的是研究發展一高速及時的 編解碼軟體系統。 從以下三方面來改善效率 1. 程式技巧 2. 演算 法 3. 容許損失些微的音訊品質經優化後, 將得到在Pentium 166機器上 只佔大約百分之三十八CPU時間的效果, 或在Pentium Pro 180機器上只佔 大約百分之二十三CPU時間的效果。 G.723.1 is a very important audio coding standard for voice.Most important video conferencing standards, such as H.323 and H.324, use G.723.1 as the audio coding standard. The reasons why G.723.1 is so important are as follows: 1. Its transmission rates are very low, 5.3 kbps and 6.3 kbps. 2. Its delay is very low, about 37.5 msec. 3. It can maintain high audio quality like telephone quality. In a quality benchmark test, it scores 3.98, while a perfect telephone scores 4.00. 4. It is robust when some packets are lost. Based on the above reasons, the goal of this thesis is to study and develop a high-speed and real-time software encoder/decoder. Three ways will improve the performance are as follows.1. Programming techniques.2. Algorithms.3. Allowing little loss of quality.From the experiments of this study, we obtain that a voice streamconsumes about 38% of CPU time on Pentium 166 and about 23% of CPU time on Pentium Pro 180.