臺灣博碩士論文加值系統

由於互連網路的發達，在網路上即時傳送影像及聲音已經成為了一種趨勢。
串流技術(streaming)就是這種一邊下載一邊播放的技術。網路傳輸訊號時難免會有遺失或是錯誤。對「先下載完再播放」的傳統技術而言，遺失握錯誤的封包，只需重新傳輸即可。但是串流技術為了因應即時播放的邀求，沒有時間重新傳輸遺失或錯誤的封包。在聲音方面，過去有人提出用重複上一個收到的片段來填補遺失或錯誤片段的方法。後來有人提出樣本比對(pattern matching)來改進它。本文作者利用mp3或aac聲音壓縮分頻段的特性，做多頻段樣本比對以填補遺失或錯誤的片段，並將重建的信號與原始訊號比對。結果顯示，多頻段樣本比對在低雜訊的情形下比不上舊有的樣本比對，但是在較高color noise時有較佳的效果。

It has been a tending to real-time transmit audio-video signals on Internet. Streaming is the technology of playing while downloading. The packet loss and crack are unavoidable in Internet transmission. For the conventional technology of play-after-download, the lost or cracked packets can simply retransmitted. However, there is no time to retransmit them in streaming. In the audio area, it was proposed to replace them by the last received frame. It was then proposed the method of pattern matching to improve. In this thesis, it was proposed further the multi-band pattern matching to make use the multi-band compression of the mp3 and aac audio. The reconstructed signals was then compared wth the original signals. It was found that the results of multi-band patt\ern matching are worse the those of the “old” pattern matching when the noise levels are low, themulti-band pattern matching has better when the color noise are higher.

摘要 i
Abstract ii
謝誌 iii
目錄 iv
圖列表 vi
表列表 viii
1.簡介 - 1 -
1.1 背景 - 1 -
1.2 動機 - 1 -
1.3 研究目標 - 2 -
1.4 大綱 - 2 -
2.串流技術 - 2 -
2.1 Real-time Transport Protocol (RTP) - 2 -
2.1.1 RTP封包格式 - 2 -
2.1.2 RTP 範例 - 4 -
2.2 RTCP - 5 -
2.2.1 RTCP 封包形態 - 5 -
2.2.2 RTCP Sending Rules - 6 -
2.2.3 RTCP封包格式 - 6 -
3.聲音壓縮 - 16 -
3.1聲音壓縮的基本 - 16 -
3.1.1靜音門檻曲線(absolute threshold)： - 16 -
3.1.2臨界頻帶(Critical band)： - 16 -
3.1.3遮蔽效應(masking effect)： - 17 -
3.2 MPEG-1 Audio layer3(Mp3)壓縮 - 17 -
3.2.1多相濾波器(Polyphase filter bank) - 18 -
3.2.2 Modified discrete cosine transform(MDCT) - 19 -
3.2.3計算遮蔽曲線 - 21 -
3.3 位元分配(Bit allocation) - 26 -
3.3.1量化 - 26 -
3.3.2 計算適當的g與cq - 26 -
4.聲音修補 - 27 -
4.1 直接重複上一個片段 - 27 -
4.2 樣本比對(pattern matching) - 27 -
4.3 多頻帶的樣本比對 - 29 -
5.語音傳輸系統 - 29 -
5.1語音傳輸系統的架構 - 29 -
5.1.1 伺服器部份 - 29 -
5.2.1 伺服端 - 30 -
5.1.2 客戶端部份 - 31 -
5.2聲音的抓取與播放 - 32 -
5.2.1聲音抓取用的函式 - 33 -
5.2.2聲音輸出用的函式 - 33 -
5.2.3 WAVEFORMATEX格式 - 34 -
5.3 聲音的傳輸 - 35 -
5.3.1 伺服端 - 35 -
5.3.2 客戶端 - 35 -
5.5封包遺失的處理 - 38 -
5.5.1 完整的聲音片段樣本比對 - 38 -
5.5.2 32個頻帶的樣本比對 - 40 -
5.5.3 兩階段的多頻帶比對 - 40 -
5.5.4 效能比較基準 - 41 -
6.系統執行結果 - 41 -
6.1 伺服端介面 - 41 -
6.2 客戶端介面 - 42 -
6.3 執行結果 - 43 -
6.4 數據 - 45 -
6.5 討論 - 63 -
7.結論 - 64 -
Reference - 64 -

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