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研究生: 陳威銘
Chen, Wei-Ming
論文名稱: HTTP線上串流即時編碼率轉換之研究與實現
Implementation of Real-Time Transrating for HTTP Live Streaming
指導教授: 黃悅民
Huang, Yueh-Min
學位類別: 碩士
Master
系所名稱: 工學院 - 工程科學系
Department of Engineering Science
論文出版年: 2012
畢業學年度: 100
語文別: 中文
論文頁數: 80
中文關鍵詞: 轉換編碼自適性串流HTTP重新導向PSNRAndroid
外文關鍵詞: transrating, adaptive streaming, HTTP redirecting, PSNR, Android
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  • 隨著智慧型手機和平板電腦等手持裝置的興起,使用者對於網路串流影片和即時串流的需求也日益增加,這樣的趨勢也間接地促進許多即時串流技術的發展,如Adobe’s RTMP、Microsoft’s Smooth Streaming以及Apple’s HTTP Live Streaming等,而傳輸方式也逐漸以通用性高的HTTP-based Streaming為主流。
    而在串流影片時期,網路頻寬發生變化往往可能會導致播放不順暢或影片延遲的現象,造成不好的使用者體驗,為了改善這種現象,自適性串流機制是一種常被用在用戶端串流時的決策方式,讓用戶端的裝置可以依據當前的網路狀況選擇不同的影片品質,不過若伺服器端提供的影片品質選擇過少,所能改善的空間就會因此受到限制。
    本文基於HTTP即時串流協定提出了一套頻寬評估機制,並且在TI PandaBoard開發版移植Google Android 4.0 Ice Crream Sandwich作業系統作為HTTP即時串流用戶端,而在伺服器端則是定義一模式轉移狀態機和三種評估失誤型態,根據頻寬評估的結果來觀察當前網路變化的情況,在不同的模式下分別採取不同的轉換編碼策略,計算合適的媒體片段位元率,將媒體片段轉換編碼為符合當前連線品質的片段,經過HTTP重新導向技術,讓HTTP即時串流用戶端能觀賞到較適切的影片品質,最後則以PNSR和頻寬使用率分析作為影片品質提升的參考指標。據實驗結果顯示,本研究於一般的網路行為可提升PSNR約2.8dB並於串流期間維持在80%以上的頻寬使用率。

    Due to the rising popularity of smart phones and tablets, the need for multimedia streaming and live video streaming increases daily on these handheld devices, causing the rapid development of live streaming protocols such as Adobe’s RTMP, Microsoft’s Smooth Streaming and Apple’s HTTP Live Streaming.
    The variations of bandwidth due to environmental or external reasons may result in unsmooth playback or interruptions of video during video playing. In order to improve quality of video streaming, adaptive streaming is proposed to be a solution for client end devices. The streaming client can choose the suitable quality of media segment when there are different qualities of video pre-encoded and stored at server. However the less the choice of quality selections provided for clients, the more limited the improvement can be made.
    In this paper, we propose a bandwidth evaluation mechanism based on HTTP Live Streaming, and port Google Android 4.0 Ice Cream Sandwich operating system on TI PandaBoard. On server side, we define a mode-transition finite state machine as well as three estimated error types. According to the result of bandwidth evaluation, we can determine the current state of network, and compute the suitable bitrate of media segment by different methods in different modes. Then the client of HTTP Live Streaming can receive a segment with the more suitable bitrate than that of the original mechanism of HTTP Live Streaming via the technique of HTTP redirecting after transrating. Finally, we analyze the PSNR and bandwidth usage as the performance metrics of quality of video streaming, and the result shows that the video quality of HTTP live streaming is improved by using our approach.

    目錄 摘要 III Abstract IV 誌謝 V 圖目錄 VIII 表目錄 XII 第一章 緒論 1 1.1 研究動機 1 1.2 研究目的 2 1.3 章節提要 3 第二章 相關標準與研究 4 2.1 RTSP (Real Time Streaming Protocol) 4 2.2 Apple’s HTTP Live Streaming 5 2.3 Transport Stream傳輸串流封裝格式 9 2.3.1 TS Packet傳輸封包 9 2.3.2 Packet Identifier傳輸編號 10 2.3.3 Program Specific Information 節目描述資訊 11 2.3.5 Program Map Table節目對應表 12 2.4 tcpdump 13 第三章 軟硬體平台介紹 15 3.1 PandaBoard硬體平台 15 3.2 Android 18 3.2.1 Android特色 18 3.2.2 Android系統架構 20 3.3 Android 多媒體框架 25 3.3.1 Android 多媒體框架演進過程 26 3.3.2 Android NuPlayer多媒體框架 30 第四章 即時轉換編碼率機制之實作 41 4.1 系統架構介紹 41 4.2 開發環境建置 43 4.2.1.伺服器端開發環境 43 4.2.2.用戶端開發環境 44 4.3 Android 應用程式 47 4.4 頻寬與影音品質之分析與改善 49 4.5 不連續轉換編碼時間戳 53 4.6 即時轉換編碼率決策機制 55 4.7 HTTP重新導向 60 第五章 系統實作結果分析 62 5.1. 測試環境 62 5.2. 串流裝置頻寬使用率分析 65 5.3. 使用者觀賞品質分析 70 第六章 結論與未來展望 75 附 錄 76 參考文獻 77

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