本文選題：實時流媒體　切入點：音視頻同步　出處：《浙江工業(yè)大學(xué)》2014年碩士論文　論文類型：學(xué)位論文

【摘要】：進入21世紀,伴隨著經(jīng)濟的高速發(fā)展,整個社會的不安定因素逐漸累積,人們的安全意識也是逐漸加強。而計算機技術(shù)、高壓縮比的數(shù)據(jù)壓縮技術(shù)以及寬帶通信技術(shù)的發(fā)展,為安防等實時流媒體系統(tǒng)應(yīng)用領(lǐng)域的發(fā)展帶來了新的增長點。隨著應(yīng)用的多樣化,單一的視頻監(jiān)控顯得有些單調(diào),已逐漸滿足不了客戶的需求,無論是銀行、鐵路、客運站,都需要優(yōu)質(zhì)的音視頻監(jiān)控系統(tǒng)提供較全面的安防服務(wù)。由于網(wǎng)絡(luò)傳輸?shù)牟豢煽啃?導(dǎo)致本應(yīng)保持同步關(guān)系的數(shù)據(jù),到達信宿端時已失去了其固有的時間關(guān)聯(lián)性,即失去了同步,因此音視頻的同步性能就成為了實時流媒體系統(tǒng)中的一個聚焦熱點。為此,本文就實時流媒體系統(tǒng)中音視頻媒體流的同步問題做了探究。設(shè)計并實現(xiàn)一個簡易的實時流媒體系統(tǒng),該系統(tǒng)包括信源端、流媒體服務(wù)器、信宿端三部分。針對音視頻數(shù)據(jù)在傳輸過程中產(chǎn)生異步的情況,設(shè)計了一個基于RTP/RTCP協(xié)議的網(wǎng)絡(luò)自適應(yīng)控制的實時流媒體同步方案。該方案從信源端、信宿端、網(wǎng)絡(luò)狀況三方面入手對音視頻實施同步控制。在信源端實現(xiàn)了音視頻數(shù)據(jù)的同步采集、編碼；音視頻數(shù)據(jù)在發(fā)送前通過打上時間戳,實現(xiàn)不同媒體映射到同一條時間軸。在信宿端,第一步是對收到的音視頻包進行同步預(yù)處理,消除時延抖動導(dǎo)致的亂序。然后分別對音頻進行解壓,對視頻進行組幀。最終依據(jù)時間戳值對音視頻數(shù)據(jù)實施同步控制。在網(wǎng)絡(luò)狀況的自適應(yīng)方面,信宿端統(tǒng)計網(wǎng)絡(luò)延時抖動與丟包,一方面依據(jù)統(tǒng)計信息調(diào)節(jié)信宿端緩存和播放速率,另一方面將統(tǒng)計信息反饋給信源端。信源端再依據(jù)接收的信息調(diào)節(jié)視頻壓縮率,實現(xiàn)發(fā)送端的碼率調(diào)節(jié)。最后經(jīng)過實驗證明,本實時流媒體系統(tǒng)在正常的網(wǎng)絡(luò)環(huán)境下具備良好的音視頻同步性能。
[Abstract]:In 21th century, with the rapid development of economy, the unstable factors of the whole society gradually accumulated, and people's security consciousness was gradually strengthened, while the development of computer technology, high compression ratio data compression technology and broadband communication technology, Along with the diversification of application, the single video surveillance appears to be a bit monotonous, and can not meet the needs of customers, whether it is the bank, railway, passenger station, and so on, which brings a new growth point for the development of real-time streaming media system, such as security and defense. Because of the unreliability of the network transmission, the data which should have kept synchronous relation have lost its inherent time correlation, that is to say, the synchronization has already been lost when arriving at the terminal of the destination, because of the unreliability of the network transmission, because of the high quality audio and video surveillance system to provide the more comprehensive security service. Therefore, the synchronization performance of audio and video has become a hot spot in real-time streaming media system. This paper probes into the synchronization of audio and video media flow in real-time streaming media system, and designs and implements a simple real-time streaming media system, which includes the source end, streaming media server, and so on. Aiming at the asynchronous situation of audio and video data transmission, a real-time streaming media synchronization scheme based on network adaptive control based on RTP/RTCP protocol is designed. The synchronization control of audio and video is implemented in three aspects of network status. The synchronous acquisition and coding of audio and video data are realized at the source end, and the audio and video data are timestamped before transmission. The first step is to synchronize the received audio and video packets to eliminate the disorder caused by the delay jitter, and then decompress the audio frequency separately. Finally, according to the time stamp value, the audio and video data are controlled synchronously. In the adaptive aspect of the network condition, the terminal counts the network delay jitter and packet loss. On the one hand, the buffer and playback rate is adjusted according to the statistical information, on the other hand, the statistical information is fed back to the source end. The source side adjusts the video compression ratio according to the received information, and realizes the rate adjustment of the sending side. Finally, the experiment proves that, The real-time streaming media system has good audio and video synchronization performance under the normal network environment.
【學(xué)位授予單位】：浙江工業(yè)大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TN919.8

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本文編號：1614308