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研究生:林宜瑋
研究生(外文):Yi-Wei Lin
論文名稱:無線行動網路上之即時視訊音訊串流機制與其應用
論文名稱(外文):Real-time Video/Audio Streaming Mechanisms and Applications overWireless/Mobile Networks
指導教授:黃崇明黃崇明引用關係
指導教授(外文):Chung-Ming Huang
學位類別:碩士
校院名稱:國立成功大學
系所名稱:資訊工程學系碩博士班
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2002
畢業學年度:90
語文別:中文
論文頁數:57
中文關鍵詞:無線網路即時媒體串流錯誤控制流量控制
外文關鍵詞:wireless networksreal-time media streamingerror controlflow control
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  • 下載下載:210
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無線網路和有線網路存在著許多不同的差異性,其中 (1) 較小的頻寬和 (2) 不穩定的傳輸媒介,使得無線網路上的即時媒體串流機制並不是那麼容易的實現。在本論文中,我們提出來一套可以在無線網路上表現平順的影音串流機制。針對音訊方面,我們藉由嵌入不同資訊的方式定義出兩種不同的傳送模式,分別為 (1) “多餘(Redundant)”模式和 (2) “重複(Duplicated)”模式。假設第i個封包中含有三個不同音訊資料,分別為i, i-1, 和i-2。在”多餘”模式下,i音訊資料利用音質較i-1和i-2音訊資料好的壓縮格式來壓縮。而在”重複”模式中,則此三個音訊資料都利用相同的壓縮格式來壓縮。如此一來,”多餘”模式可以給予較好的音訊品質;但也同時需要耗損掉較大的頻寬。針對視訊方面,我們利用視訊分層的技術來達成視訊傳送率的調整。兩個網路資訊, (1) 封包遺失率(packet loss rate)和 (2) 來回傳遞時間(round-trip time),被用來偵測網路狀態。兩個臨界點被設定出來,分別為 (1) 上臨界點和 (2) 下臨界點。當平均來回傳遞時間超過最大來回傳遞時間的上臨界點時,網路進入了壅塞(congested)狀態;當平均來回傳遞時間低於最大來回傳遞時間的下臨界點時,網路進入了空蕩(unloaded)狀態。由於無線網路上的傳遞媒介不如有線網路那般穩定,並非所有的封包遺失都是因為網路壅塞所導致。所以在本論文中,我們引用了一個分隔封包遺失特性的方法分辨出是否目前的封包遺失代表著網路壅塞。此方法使用相隔來到時間(inter-arrival time)來判別是否一個不依順序到來的封包是即時進入接收端。如果該封包是即時進入接收端的話,遺失掉的封包即被視為因不穩定媒介而遺失;如果該封包不是即時進入,則該封包為網路壅塞所導致的遺失。當網路壅塞時,傳送端必須降低傳送速率;當網路空蕩時,傳送端則可提高傳送速率。藉由此方法,在本論文中我們完成了一個可以在無線網路上依網路情況調變傳送速率的即時影音串流機制。
Due to the characteristics of (1) smaller bandwidth and (2) unreliable transmission media, real-time media streaming over wireless networks is not trivial. To have smooth media streaming over wireless networks, we propose schemes for video and audio streaming respectively. For audio, two sending modes, in which redundant information is embedded in each packet, that the proposed scheme contains are (1) the "redundant" mode and (2) the "duplicated" mode. Let a packet i can contain three audio frames i, i-1, and i-2. In the redundant mode, frame i uses a codec of better quality than that for frames i-1 and i-2. In the duplicated mode, frames i, i-1, and i-2 use the same codec, which has lower quality than that for frame i used in the redundant mode. The "redundant" mode may give better quality of sound but consumes more bandwidth, while the "duplicated" mode gives lower quality of sound but consumes less bandwidth. For video, an adaptive real-time video streaming scheme that uses the layered video technique is proposed. Two attributes that are used to determine the network situation and then adjust the sending rate accordingly are loss-rate and round-trip time (RTT). Two thresholds named "upper-ratio" and "lower-ratio" are set. When the average RTT exceeds the upper-ratio multiplies the maximum RTT, the network situation is set to congested; when the average RTT is under the lower-ratio multiplies the maximum RTT, the network situation is set to unloaded. Since the unreliable media cause of packet loss in the wireless environment is rate-independent, a method that can separate these rate-independent loss from the congestion loss is needed. We use inter-arrival time between two received packets to identify if an out of order packet was received in time. If the out of order packet was received in time, then the packet loss between the two received packets is caused by unreliable media; otherwise, the packet loss is caused by congestion and the network situation is congested. Upon a congestion situation is determined, the sending rate of the sender is dropped down; upon unloaded situation is determined, the sending rate of the sender is raised up. In this way, the adaptive real-time media streaming can be achieved for wireless networks.
1 Introduction ................................................ 1
2 Preliminary ................................................ 4
2.1 Error Control .................................... 4
2.2 Flow Control .................................... 7
3 System Architecture .........................................10
3.1 Functions of Wireless Gateway ....................11
3.2 System Components ................................14
4 The Proposed Scheme .........................................17
4.1 Synchronization between Video and Audio ..........17
4.2 Error Control for Audio ..........................18
4.3 Flow Control .....................................20
5 Performance Evaluation ......................................29
5.1 Audio Part Evaluation ............................29
5.2 Video Part Evaluation ............................38
5.2.1 3G Cellular Network Evaluation ........38
5.2.2 Wireless LAN Environment Evaluation ...48
6 Conclusion ..................................................53
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