非連續語音傳輸系統

Abstract

本論文之製作-非連續語音傳輸系統，主要探討如何節省話務負荷 以及 舒解網路擁塞的可行性。首先藉著對包覆式增量調變編碼的理論分析確定 這個架構對傳輸誤差的忍受程度，繼而研究其執行成效，藉以改善提高輸 出音質。同時我們也針對雙向通話叢發特性加入音態鑑別以節省話務負荷 ，並探討使用數位信號處理器製作之可行性。有關音態鑑別器的設計主要 是採用傳輸位元變化率和預估增益，這兩項指標參數在靜音或有聲音態時 之統計分布特性差異，並據以設定判別區間的臨界值。至於接收端的雜訊 補償，其作用在能有效模擬靜音時中斷傳送的背景噪音聲效。若將雜訊的 振幅及模型參數視為附加訊息傳送，接收器便可據以製作一補償濾波器， 並配合隨機變數的激發以合成虛擬的背景噪音。 This work is aimed to benefit from the concept of discontinuous transmission by sending the DM-encoded speech packets only during talkspurts. We first analyze the effect of transmission errors for EDM coding of autoregressive sources, and we substantiate the theoretical values with simulation values. The combined use of bit alterating rate and prediction gain allows the implementation of a voice activity detector that accurately distinguish between active and silent periods. Furthermore, a comfort noise generator is used to produce on the receiver side synthetic background noise which is not transmitted during the silent periods. This task is done by using the random noise source to excite an all-pole synthesis filter which is characterized by both the short-term energy and the linear prediction parameters of background noise signal.