臺灣博碩士論文加值系統

人性化隨身資訊服務是未來的發展趨勢，其關鍵在於開發一聲控操作的語音人機介面。本論文在網際網路環境，建構一分散式語音辨認系統，再根據辨認結果回傳特定的有聲資訊給用戶。網路語音通訊最重要的課題是服務品質管理，特別是封包漏失、傳輸延遲及延遲顫動。為了補償封包漏失，我們採用多重敘述編碼架構，透過兩個獨立的網路通道傳送語音封包。至於延遲擾動的解決方案，一般是在接收端加入一播放緩衝器暫存語音封包，再彈性調整每個語音封包的播放時間。由於網路延遲在話務中間的變動，語音封包的晚到漏失率與其緩衝延遲及之間存在一個最佳化權衡的問題。我們將在多重敘述編碼架構下，根據客觀的音質預估模型，針對每個獨立封包的播放延遲進行音質最佳化調整。

The purpose of this research is to develop an interactive voice response system that allows drivers to use voice-controlled commands to access the information server through the internet. We first implement a distributed speech recognition system, in which speech features extracted from a local front-end are transmitted through a data channel to a remote back-end recognition server. Another important issue to address is the playout buffer design, which is often used at the receiver to smooth out the jitter for timely reconstruction of the speech. We formulate the adaptive playout scheduling of multiple voice streams as a constrained optimization problem that leads to a better balance between end-to-end delay and packet loss. Also proposed is a perceptually motivated optimization criterion and a practically feasible algorithm for the playout buffer design.

第一章 緒論 1
1.1 研究動機與方向 1
1.2 章節概要 2
第二章 語音人機介面 3
2.1系統介面設計 4
2.1.1分散式語音辨識 5
2.1.2軟體架構 7
2.1.3 網路協定封包製作 7
2.1.4軟體運作的設定 10
2.1.5 檔案架構 11
2.2 MANET網路的延遲時間量測分析 14
第三章 播放排程演算法 25
3.1 播放緩衝器簡介 26
3.2播放緩衝器效能分析 30
3.3 適應性播放演算法 32
3.4多重敘述編碼架構 35
第四章 播放排程的聽覺最佳化設計 39
4.1通話品質預測模型 40
4.1.1主觀聽覺測試 40
4.1.2 音質評量指標 42
4.2音質最佳化的播放排程機制 47
4.2.1 音質最佳化的設計 47
4.2.2 緩衝漏失機率模型 49
4.3 安全因子的動態調整機制 52
4.4 最佳化的割線演算法 53
第五章 實驗結果 57
5.1 網路單向延遲模型 57
5.1.1 模型簡介 57
5.1.2 網路延遲分析 60
5.2 移動式自組網路下的封包傳輸延遲 63
5.3 播放排程演算法的效能比較 68
第六章 結論與未來展望 71
參考文獻 72

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