發(fā)布時(shí)間：2018-03-28 19:17

  本文選題：語音編碼　切入點(diǎn)：電話信令　出處：《哈爾濱工業(yè)大學(xué)》2014年碩士論文

【摘要】：隨著科學(xué)技術(shù)的進(jìn)步,電子產(chǎn)品的成本和價(jià)格不斷下調(diào),智能終端已經(jīng)深入了人們的生活,成為了日常工作學(xué)習(xí)中不可缺少的工具,而Android系統(tǒng)依托開放共贏的理念以系統(tǒng)開源的形式快速地占領(lǐng)了智能終端市場的絕大份額,地位越來越重要。針對Android系統(tǒng)良好的應(yīng)用前景,本課題將Android平臺(tái)作為實(shí)現(xiàn)平臺(tái)。近些年,電話信令技術(shù)、網(wǎng)絡(luò)技術(shù)、多媒體技術(shù)、語音編解碼技術(shù)、網(wǎng)絡(luò)穿透技術(shù)不斷進(jìn)步,同時(shí)互聯(lián)網(wǎng)應(yīng)用迅速興起,網(wǎng)絡(luò)電話在兼有良好技術(shù)基礎(chǔ)及廣闊的市場應(yīng)用前景下得到了快速的發(fā)展,在工作生活中越來越多的開始取代傳統(tǒng)電信網(wǎng)絡(luò)通信。本文的研究內(nèi)容便為基于Android的網(wǎng)絡(luò)語音通話軟件設(shè)計(jì)。開發(fā)出具有實(shí)用價(jià)值的通信軟件為實(shí)驗(yàn)室后續(xù)在智能家居領(lǐng)域的進(jìn)一步拓展具有實(shí)質(zhì)性的意義。系統(tǒng)由客戶端和服務(wù)器兩部分組成�？蛻舳酥饕ㄕZ音信號(hào)處理及傳輸、電話信令SIP、NAT網(wǎng)絡(luò)穿透三個(gè)模塊,服務(wù)器主要完成用戶上線注冊、用戶之間尋址以及用戶注冊的功能。本文工作的主要內(nèi)容如下:設(shè)計(jì)實(shí)現(xiàn)了語音信號(hào)處理及傳輸過程。包括語音的采集、編碼、發(fā)送、接收、解碼及播放六個(gè)部分,每部分使用獨(dú)立的線程完成。首先調(diào)用Android平臺(tái)對語音信號(hào)進(jìn)行采集,將采集到的語音信號(hào)流交給ILBC編碼庫進(jìn)行編碼,對編碼完成的語音數(shù)據(jù)即時(shí)交給發(fā)送線程使用socket技術(shù)傳輸給指定地址,接收線程則通過socket技術(shù)監(jiān)聽指定端口號(hào)接收語音數(shù)據(jù),并將接收的數(shù)據(jù)即時(shí)交給解碼線程進(jìn)行解碼,播放線程將已解碼數(shù)據(jù)即時(shí)播放。設(shè)計(jì)實(shí)現(xiàn)了SIP信令客戶端。使用Android系統(tǒng)自帶的API進(jìn)行SIP信令客戶端的開發(fā),客戶端主要實(shí)現(xiàn)有用戶信息注冊及保存、通話的發(fā)起、來電接收的基本功能。設(shè)計(jì)實(shí)現(xiàn)了SIP基本功能服務(wù)器。使用開源服務(wù)器代碼架設(shè)了用于SIP通話的服務(wù)器,基于該服務(wù)器開發(fā)了用戶管理的軟件界面,實(shí)現(xiàn)了對用戶簡易注冊及刪除的基本管理功能。經(jīng)過實(shí)際測試,良好的實(shí)現(xiàn)了語音信號(hào)的處理及傳輸過程,在局域網(wǎng)之內(nèi)語音通話質(zhì)量清晰且延時(shí)感不明顯,SIP客戶端實(shí)現(xiàn)了SIP服務(wù)器的登錄,能夠給指定SIP用戶建立SIP通話,服務(wù)器實(shí)現(xiàn)了對用戶的注冊及注銷過程。
[Abstract]:With the progress of science and technology, the cost and price of electronic products have been continuously reduced. Intelligent terminals have become an indispensable tool in daily work and learning. The Android system, relying on the concept of open and win-win, has occupied the vast majority of the intelligent terminal market rapidly in the form of open source system, and its status is becoming more and more important. In view of the good application prospects of Android system, In recent years, the telephone signaling technology, network technology, multimedia technology, voice coding and decoding technology, network penetration technology, the rapid rise of Internet applications, Internet telephony has developed rapidly with both good technical foundation and broad market application prospects. More and more people begin to replace the traditional telecommunication network communication in the work life. The research content of this paper is the design of the network voice communication software based on Android. The system consists of two parts: the client and the server. The client mainly includes voice signal processing and transmission. Telephone signaling SIPN Nat network penetrates three modules, the server mainly completes the subscriber on-line registration. The main contents of this paper are as follows: the design and implementation of speech signal processing and transmission process, including voice acquisition, coding, sending, receiving, decoding and playing six parts, Each part is completed by independent thread. Firstly, the Android platform is called to collect the speech signal, and the stream of the collected speech signal is given to the ILBC coding library for coding. On the other hand, the encoded voice data is immediately transferred to the sending thread using socket technology to receive the specified address, while the receiving thread listens to the specified port number to receive the voice data through socket technology, and the received data is immediately handed over to the decoding thread for decoding. The playback thread plays the decoded data instantly. The SIP signaling client is designed and implemented. The SIP signaling client is developed by using the API which comes with the Android system. The client mainly realizes the registration and saving of the user information and the initiation of the call. The basic function of receiving calls. The basic function server of SIP is designed and implemented. The server for SIP calls is set up by using open source server code, and the software interface of user management is developed based on this server. The basic management function of simple registration and deletion of users is realized. After practical test, the process of speech signal processing and transmission is well realized. In the LAN, the voice call quality is clear and the delay sense is not obvious. The client realizes the login of the SIP server, which can set up the SIP call for the designated SIP user, and the server realizes the process of registering and deregistration the user.
【學(xué)位授予單位】：哈爾濱工業(yè)大學(xué)
【學(xué)位級(jí)別】：碩士
【學(xué)位授予年份】：2014
【分類號(hào)】：TN916.5;TP311.52

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本文編號(hào)：1677704